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<contents><![CDATA[If you are unable to get your Linksys PAP2T-NA Analog Telephone Adapter to ring when calling your handset change the following setting on the PAP2T and you should be good to go:<br />
<br />
Regional tab (advanced view)<br />
Ring and Call Waiting Tone Spec > Ring Waveform: Sinusoid to Trapezoid<br />
<br />
Default is Sinusoid which needs to be changed to Trapezoid.<br />
<br />
Also you can download the latest firmware (3.1.13 as of the time of this post) in our Downloads section.<br />
http://www.888voipstore.com/support/index.php?_m=downloads&_a=view<br />
]]></contents>
<contentstext><![CDATA[If you are unable to get your Linksys PAP2T-NA Analog Telephone Adapter to ring when calling your handset change the following setting on the PAP2T and you should be good to go:

Regional tab (advanced view)
Ring and Call Waiting Tone Spec > Ring Waveform: Sinusoid to Trapezoid

Default is Sinusoid which needs to be changed to Trapezoid.

Also you can download the latest firmware (3.1.13 as of the time of this post) in our Downloads section.
http://www.888voipstore.com/support/index.php?_m=downloads&_a=view
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Linksys PAP2T-NA won't ring a handset]]></subject>
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<contents><![CDATA[Wire instructions:<br />
<br />
Bank of America
900 W. Trade St 
Charlotte, NC 28255
Phone: 800-360-9480 

ACCT: 229016565828
Swift: BOFAUS3N (If Needed)
Inbound Wiring Routing ABA#: 026009593
]]></contents>
<contentstext><![CDATA[Wire instructions: 
 
Bank of America
900 W. Trade St 
Charlotte, NC 28255
Phone: 800-360-9480 

ACCT: 229016565828
Swift: BOFAUS3N (If Needed)
Inbound Wiring Routing ABA#: 026009593
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[888VoipStore Wiring Instructions]]></subject>
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<contents><![CDATA[Question:<br />
<br />
"I have an office at home and I would like to connect my phone and Fax machine to two of my Analog phones at the office. I have a cable modem at home and a T1 connection at the office. Would you please tell me first if this is possible and if yes what do I need to buy and what is the best brand to buy. I heard Grandstream is a good brand. Is that true?"<br />
<br />
Response:<br />
<br />
For your home you can use an analog adapter with 2 ports - 1 for fax & 1 for phone.  I recommend the <a href="http://www.888voipstore.com/linksys-pap2t-na-pr-16310.html" target="_blank">PAP2T</a>.<br />
<br />
You will need a voip service provider to give you a local telephone # if you wish to give this out for "non-voip" users to use with regular telephony equipment.  We don't currently recommend a specific provider as there may be different alternatives that best suit your needs.  The only required feature of a VoIP service provider is that you can "use your own hardware" - and that "they will help you provision said hardware".<br />
<br />
For your Office with the T-1 - You can choose a Grandstream <a href="http://www.888voipstore.com/grandstream-gxp-2000-pr-16165.html" target="_blank">GXP-2000</a>.<br />
<br />
Yes Grandstream is a good brand and is cost effective.<br />
<br />
<br />
888VoipStore<br />
]]></contents>
<contentstext><![CDATA[Question: 
 
"I have an office at home and I would like to connect my phone and Fax machine to two of my Analog phones at the office. I have a cable modem at home and a T1 connection at the office. Would you please tell me first if this is possible and if yes what do I need to buy and what is the best brand to buy. I heard Grandstream is a good brand. Is that true?" 
 
Response: 
 
For your home you can use an analog adapter with 2 ports - 1 for fax & 1 for phone.  I recommend the PAP2T. 
 
You will need a voip service provider to give you a local telephone # if you wish to give this out for "non-voip" users to use with regular telephony equipment.  We don't currently recommend a specific provider as there may be different alternatives that best suit your needs.  The only required feature of a VoIP service provider is that you can "use your own hardware" - and that "they will help you provision said hardware". 
 
For your Office with the T-1 - You can choose a Grandstream GXP-2000. 
 
Yes Grandstream is a good brand and is cost effective. 
 
 
888VoipStore 
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Connecting your home to your office with VoIP]]></subject>
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<contents><![CDATA[How to locate your PAP2 / PAP2T IP address & access the Configuration GUI<br />
<br />
1.- Connect a telepohone to the phone port<br />
<br />
2.- Press **** 110 #<br />
<br />
3.- Enter on http://localip where localip is gotten via **** 110 # on telephone <br />
<br />
4.- Press Admin Login.]]></contents>
<contentstext><![CDATA[How to locate your PAP2 / PAP2T IP address & access the Configuration GUI

1.- Connect a telepohone to the phone port

2.- Press **** 110 #

3.- Enter on http://localip where localip is gotten via **** 110 # on telephone 

4.- Press Admin Login.]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[How to locate your PAP2 / PAP2T IP address & access the Configuration GUI]]></subject>
<isedited><![CDATA[1]]></isedited>
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<contents><![CDATA[Q: Why is there a humming noise when using a headset on a VoIP phone?<br />
<br />
A: That is a common problem on VoIP phones due to a missing ground connection. The noise is caused by the metal microphone tube and a metal part of the headset, respectively, which act as antennas. There are two simple solutions:<br />
<br />
   1. Connect the phone to the switch with a shielded Ethernet cable + connector (make sure your wall socket/switch/hub has the shielded socket connected to the ground; most of the switches available on the market do).<br />
<br />
   2. In case your network environment is not grounded you can use the phone as a switch and connect the second port of the phone (PC port)to the Ethernet card of the PC by means of a shielded Ethernet cable. The PC card is grounded by default. ]]></contents>
<contentstext><![CDATA[Q: Why is there a humming noise when using a headset on a VoIP phone?

A: That is a common problem on VoIP phones due to a missing ground connection. The noise is caused by the metal microphone tube and a metal part of the headset, respectively, which act as antennas. There are two simple solutions:

   1. Connect the phone to the switch with a shielded Ethernet cable + connector (make sure your wall socket/switch/hub has the shielded socket connected to the ground; most of the switches available on the market do).

   2. In case your network environment is not grounded you can use the phone as a switch and connect the second port of the phone (PC port)to the Ethernet card of the PC by means of a shielded Ethernet cable. The PC card is grounded by default. ]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Why is there a humming noise when using a headset on a VoIP phone?]]></subject>
<isedited><![CDATA[1]]></isedited>
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<contents><![CDATA[-------------------------------------------------------------------<br />
     To create a product Return Request:<br />
-------------------------------------------------------------------<br />
<br />
1. Log into your account at http://www.888voipstore.com<br />
<br />
   Note: If you lost your password click on "Recover Password"<br />
   and enter your email address.  Passwords can only be sent<br />
   to the email address registered when the account was created.<br />
   Please contact support@888voipstore.com or call 888-864-7786 x3<br />
   if you are having difficulty logging into your account.<br />
<br />
2. Select "Orders history" from "Your cart" menu on the right.<br />
<br />
3. Search for the order you want to create a Return Request for.<br />
<br />
4. Click on its number or status, and the Order details page opens.<br />
<br />
5. Scroll down to the Create Return Request form.<br />
<br />
   Note: You may only create Return Requests for products within<br />
   the product return time of 30 days from receipt of Mechandise.<br />
   If you do not see the Create Return Request form on the Order<br />
   details page, the time during which the Return Request could<br />
   be placed has expired.  Please contact support@888voipstore.com<br />
   or call 888-864-7786 x3 for further options.<br />
<br />
6. Set the amount of the product(s) you want to return.<br />
<br />
7. Select the reason for returning.<br />
<br />
8. Select the action you would like 888VoipStore to do for you.<br />
<br />
9. Enter the serial #'s of the items you are requesting to be<br />
   returned. If the item is defective, please type a detailed<br />
   description of the problem including any case #'s you may have<br />
   received from the manufacturer.<br />
<br />
10. Click on Create.<br />
<br />
After the Return Request has been successfully created, the<br />
confirming notification appears at the top of the page.<br />
<br />
After a Return Request is approved you will receive an RMA # along<br />
with instructions on how to ship your return back to us.<br />
<br />
If at any time you need assistance with your Return Request,<br />
please email support@888voipstore.com or call 888-864-7786 x3<br />
<br />
-------------------------------------------------------------------<br />
     Managing previously placed Return Requests<br />
-------------------------------------------------------------------<br />
<br />
You can modify/delete the Return Requests previously placed by you<br />
via the Returns item in Your cart menu.<br />
<br />
To modify an existing Return Request:<br />
<br />
1. Click on Returns in Your cart menu.<br />
<br />
2. Use the Returns search form to find the Return Request you would<br />
   like to modify. The search results will be displayed.<br />
<br />
3. Do one of the following:<br />
<br />
- Select the check box next to the Return Request you would like to<br />
modify and click on Modify.<br />
<br />
- Click on the return ID link of the request you would like to<br />
  modify. The return Modify form opens.<br />
<br />
4. Select the reason for returning the product.<br />
<br />
5. Select an action from the drop-down list.<br />
<br />
6. Enter the serial #'s of the items you are requesting to be<br />
   returned. If the item is defective, please type a detailed<br />
   description of the problem including any case #'s you may have<br />
   received from the manufacturer.<br />
<br />
7. Click on Modify.<br />
<br />
-------------------------------------------------------------------<br />
     To delete an existing Return Request:<br />
-------------------------------------------------------------------<br />
<br />
1. Search for the necessary Return Request in the Returns section.<br />
<br />
2. In the Returns list, select the check box next to the request<br />
   you would like to remove.<br />
<br />
3. Click on Delete selected.]]></contents>
<contentstext><![CDATA[-------------------------------------------------------------------
     To create a product Return Request:
-------------------------------------------------------------------

1. Log into your account at http://www.888voipstore.com

   Note: If you lost your password click on "Recover Password"
   and enter your email address.  Passwords can only be sent
   to the email address registered when the account was created.
   Please contact support@888voipstore.com or call 888-864-7786 x3
   if you are having difficulty logging into your account.

2. Select "Orders history" from "Your cart" menu on the right.

3. Search for the order you want to create a Return Request for.

4. Click on its number or status, and the Order details page opens.

5. Scroll down to the Create Return Request form.

   Note: You may only create Return Requests for products within
   the product return time of 30 days from receipt of Mechandise.
   If you do not see the Create Return Request form on the Order
   details page, the time during which the Return Request could
   be placed has expired.  Please contact support@888voipstore.com
   or call 888-864-7786 x3 for further options.

6. Set the amount of the product(s) you want to return.

7. Select the reason for returning.

8. Select the action you would like 888VoipStore to do for you.

9. Enter the serial #'s of the items you are requesting to be
   returned. If the item is defective, please type a detailed
   description of the problem including any case #'s you may have
   received from the manufacturer.

10. Click on Create.

After the Return Request has been successfully created, the
confirming notification appears at the top of the page.

After a Return Request is approved you will receive an RMA # along
with instructions on how to ship your return back to us.

If at any time you need assistance with your Return Request,
please email support@888voipstore.com or call 888-864-7786 x3

-------------------------------------------------------------------
     Managing previously placed Return Requests
-------------------------------------------------------------------

You can modify/delete the Return Requests previously placed by you
via the Returns item in Your cart menu.

To modify an existing Return Request:

1. Click on Returns in Your cart menu.

2. Use the Returns search form to find the Return Request you would
   like to modify. The search results will be displayed.

3. Do one of the following:

- Select the check box next to the Return Request you would like to
modify and click on Modify.

- Click on the return ID link of the request you would like to
  modify. The return Modify form opens.

4. Select the reason for returning the product.

5. Select an action from the drop-down list.

6. Enter the serial #'s of the items you are requesting to be
   returned. If the item is defective, please type a detailed
   description of the problem including any case #'s you may have
   received from the manufacturer.

7. Click on Modify.

-------------------------------------------------------------------
     To delete an existing Return Request:
-------------------------------------------------------------------

1. Search for the necessary Return Request in the Returns section.

2. In the Returns list, select the check box next to the request
   you would like to remove.

3. Click on Delete selected.]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Returning products]]></subject>
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<contents><![CDATA[<p align="center"><strong>FAQ                             - Frequently Asked Questions</strong></p>
<p align="right">&nbsp;</p>
<blockquote>
  <blockquote>
    <blockquote>
      <p><a name="H" id="H"></a><strong>Handytone Questions</strong></p>
      <p><a name="H1" id="H1"></a><strong>Q. What are the main differences                                   between the 5 models of Handytones?</strong></p>
      <p>Answer:-</p>
      <div align="center">
        <table border="1" cellpadding="0" cellspacing="0" width="92%">
          <tbody>
            <tr>
              <td bgcolor="#ccccff" width="29%"><p>Features</p></td>
              <td bgcolor="#ccccff" width="15%"><p align="center"><a href="http://www.888voipstore.com/grandstream-handytone-286-pr-16190.html">HT-286</a> </p></td>
              <td bgcolor="#ccccff" width="19%"><p align="center"><a href="http://www.888voipstore.com/grandstream-handytone-386-pr-16191.html">HT-386</a></p></td>
              <td bgcolor="#ccccff" width="19%"><p align="center"><a href="http://www.888voipstore.com/grandstream-handytone-486-pr-16192.html">HT-486</a> </p></td>
              <td bgcolor="#ccccff" width="18%"><p align="center"><a href="http://www.888voipstore.com/grandstream-handytone-488-pr-16179.html">HT-488</a></p></td>
              <td bgcolor="#ccccff" width="18%"><p align="center"><a href="http://www.888voipstore.com/grandstream-handytone-496-pr-16180.html">HT-496</a></p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>Ethernet                                           Ports</p></td>
              <td><p align="center">1                                           RJ45<br />
                (LAN)</p></td>
              <td><p align="center">1                                           RJ45<br />
                (LAN) </p></td>
              <td><p align="center">2                                           RJ45<br />
                (LAN/WAN)</p></td>
              <td><p align="center">2                                           RJ45<br />
                (LAN/WAN)</p></td>
              <td><p align="center">2                                           RJ45<br />
                (LAN/WAN)</p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>DHCP/NAT/Router</p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">Yes</p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>FXS                                           Port</p></td>
              <td><p align="center">1</p></td>
              <td><p align="center">2</p></td>
              <td><p align="center">1</p></td>
              <td><p align="center">1</p></td>
              <td><p align="center">2</p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>FXO                                           Port </p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">No</p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>PSTN                                           Pass-through Port</p></td>
              <td><p align="center">No</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">Yes</p></td>
              <td><p align="center">No</p></td>
            </tr>
            <tr>
              <td bgcolor="#cccccc"><p>Remote                                           Configuration</p></td>
              <td><p align="center">TFTP/HTTP</p></td>
              <td><p align="center">TFTP/HTTP</p></td>
              <td><p align="center">TFTP/HTTP</p></td>
              <td><p align="center">TFTP/HTTP</p></td>
              <td><p align="center">TFTP/HTTP</p></td>
            </tr>
          </tbody>
        </table>
      </div>
      <p><a name="H2" id="H2"></a><strong>Q. Can you explain the use                                   of &lsquo;PSTN pass through&rsquo; and &lsquo;FXO port&rsquo;? </strong></p>
      <p>Answer:- </p>
      <p><strong>PSTN</strong><strong> Pass</strong><strong> through port:</strong><br />
        What it can do:<br />
        - Local manual switching between PSTN and IP mode                                   on a per call basis.<br />
        - User can switch to PSTN line by pressing *00                                   (or the configured strings) for each call before                                   they are placed. The device will revert back to                                   the default IP mode once the phone is hung up. <br />
        - It can allow a PSTN call to ring/call the phone                                   connected to the FXS port. <br />
        - It also serves as a life line in case of power                                   outage. <br />
        What it CANNOT do:<br />
        - Terminate a VoIP call into the PSTN port<br />
        - Allow a call from PSTN to route other VoIP devices                                   (different from the FXS phone) over the IP network<br />
        - Automatically route calls made by the local                                   user to PSTN line </p>
      <p>Note: On the HT-486 <em>Rev 1.0, </em>the PSTN                                   port is only a life line port that switches to                                   PTSN only on loss of power.</p>
      <p><strong>FXO port:</strong><br />
        It can support all the functions of a PSTN pass                                   through plus:<br />
        - Terminate a VoIP call into the PSTN port<br />
        - Allow a PSTN call to call either the FXS phone                                   or other VoIP devices over the IP network<br />
        - Route call automatically and transparently to                                   PSTN line according to user configuration </p>
      <p><a name="H3" id="H3"></a><strong>Q. Can I call from FXS1 port                                   to FXS2 port? </strong></p>
      <p>Answer:-</p>
      <p>Yes, they can communicate with each other by                                   dialing the respective extension number. Both                                   FXS ports need a valid sip account registered                                   on the server. </p>
      <p><a name="H4" id="H4"></a><strong>Q. How can I disable hook/flash?</strong></p>
      <p>Answer:-</p>
      <p>On the Advanced Settings page, there is a field                                   On-Hook Threshold. One on the selections for this                                   field is &lsquo;Hook/Flash OFF&rsquo;, this option will disable                                   hook/flash on the Phone connected to the ATA.                                   Now, in order to switch to a second channel you                                   will need to press FLASH button on the Phone,                                   instead of doing hook/flash.</p>
      <p><em>Note:- This feature is not available on older                                   hardware revision models.</em></p>
      <p><a name="H5" id="H5"></a><strong>Q. What hardware revision is                                   my Handytone?</strong></p>
      <p>Answer:-</p>
      <p>At the bottom of the device, there will be a                                   white sticker. On this sticker will be a note:</p>
      <p align="center">&nbsp;&lsquo;Rev:                                   x.0&rsquo; where x is the Hardware Revision</p>
      <p>You can also get this information under the Status                                   tab of the Web Configuration pages.</p>
      <p>Your hardware revision is very important information.                                   Depending on your hardware revision, it may/may                                   not be able to upgrade to the latest release. </p>

      <p>Ex. HT286 Rev 2.0 upgrades upto 1.0.7.19 firmware,                                   while Rev 3.0 can upgrade to the latest release.</p>
      <p><em>Note:- Once there is a new hardware revision                                   out in the market, the older revision is no more                                   manufactured or sold. </em></p>
      <p><a name="H6" id="H6"></a><strong>Q. How do I access the Web                                   Configuration pages (for HT486/496/488)?</strong></p>
      <p>Answer:-</p>
      <p>Please disconnect all connections for HT486/HT496/HT488                                   and follow the instructions below:-</p>
      <p>1. Connect the Regular Analog Touch tone phone                                   to the HT.</p>
      <p>2. Connect Power Supply.</p>
      <p>3. Now, connect the Ethernet cable between the                                   INTERNET Source (ex. Router, Modem) and the WAN                                   port on the HT.</p>
      <p>4. Now, connect another Ethernet cable between                                   your PC and the LAN port on the HT</p>
      <p>5. Wait for 30 seconds till your PC gets an IP                                   ADDRESS (192.168.2.2)</p>
      <p>6. Now, open Internet Explorer and type in 192.168.2.1,                                   you should see Grandstream Login Screen pop up.</p>
      <p>7. Enter &lsquo;admin&rsquo; as the password</p>
      <p>8. Go to Advanced Settings page and switch &quot;Enable                                   WAN port HTTP access&quot; to YES, hit &lsquo;Update&rsquo;                                   and then &lsquo;Reboot&rsquo;.</p>
      <p>9. Now, disconnect your PC from the LAN port                                   and connect it to any other port on your Router                                   within the same LAN Segment</p>
      <p>10. Now type in the actual IP ADDRESS of the                                   HT (You can look this up by pressing *** on the                                   phone, and then 02) on Internet browser, access                                   the Web Configuration page as you did earlier                                   and configure the device by filling in the information                                   given by your Internet Telephony Service Provider.</p>
      <p><a name="H7" id="H7"></a><strong>Q. How do I perform a 3-way                                   Conference call?</strong></p>
      <p>Answer:-</p>
      <p>3 way calling between HT, A and B:-</p>
      <p>1. HT calls A</p>
      <p>2. HT talks to A</p>
      <p>3. HT presses FLASH or hook/flash and gets a                                   new dialtone</p>
      <p>4. A is on Hold</p>
      <p>5. HT dials *23 and number for B</p>
      <p>6. HT talks to B</p>
      <p>7. HT presses FLASH or hook/flash to initiate                                   the 3 way calling</p>
      <p><a name="H8" id="H8"></a><strong>Q. How do I perform Attend                                   Transfer?</strong></p>
      <p>Answer:-</p>
      <p>Attend Transfer from A to B through HT:-</p>
      <p>1. A calls HT</p>
      <p>2. HT talks to A</p>
      <p>3. HT presses FLASH or does hook/flash to get                                   new dialtone.</p>
      <p>4. A is on Hold</p>
      <p>5. HT calls B</p>
      <p>6. HT talks to B</p>
      <p>7. HT hangs up to perform the Attend Transfer.</p>
      <p>8. A and B are in call now.</p>
      <p><a name="H12" id="H12"></a><strong>Q. Which Phone will ring if                                   there is an incoming call on my Fixed Line (for                                   HT386)?</strong></p>
      <p>Answer:-</p>
      <p>The Phone connected to FXS1 port will ring when                                   there is an incoming call on the Fixed Line.&nbsp; </p>
      <p><a name="H13" id="H13"></a><strong>Q. How do I make or receive                                   PSTN calls on my HT (for HT486/386/488)?</strong></p>
      <p>Answer:-</p>
      <p>For receiving a call, the Phone connected to                                   the HT will simply ring.</p>
      <p>For making a call, you will need to press the                                   &lsquo;PSTN Line Access Code&rsquo; first, as configured on                                   the Web Configuration Page (by default it is *00),                                   and then dial the desired PSTN number.</p>
      <p><br />
      </p>
    </blockquote>
  </blockquote>
</blockquote>
]]></contents>
<contentstext><![CDATA[FAQ                             - Frequently Asked Questions
&nbsp;

  
    
      Handytone Questions
      Q. What are the main differences                                   between the 5 models of Handytones?
      Answer:-
      
        
          
            
              Features
              HT-286 
              HT-386
              HT-486 
              HT-488
              HT-496
            
            
              Ethernet                                           Ports
              1                                           RJ45 
                (LAN)
              1                                           RJ45 
                (LAN) 
              2                                           RJ45 
                (LAN/WAN)
              2                                           RJ45 
                (LAN/WAN)
              2                                           RJ45 
                (LAN/WAN)
            
            
              DHCP/NAT/Router
              No
              No
              Yes
              Yes
              Yes
            
            
              FXS                                           Port
              1
              2
              1
              1
              2
            
            
              FXO                                           Port 
              No
              No
              No
              Yes
              No
            
            
              PSTN                                           Pass-through Port
              No
              Yes
              Yes
              Yes
              No
            
            
              Remote                                           Configuration
              TFTP/HTTP
              TFTP/HTTP
              TFTP/HTTP
              TFTP/HTTP
              TFTP/HTTP
            
          
        
      
      Q. Can you explain the use                                   of &lsquo;PSTN pass through&rsquo; and &lsquo;FXO port&rsquo;? 
      Answer:- 
      PSTN Pass through port: 
        What it can do: 
        - Local manual switching between PSTN and IP mode                                   on a per call basis. 
        - User can switch to PSTN line by pressing *00                                   (or the configured strings) for each call before                                   they are placed. The device will revert back to                                   the default IP mode once the phone is hung up.  
        - It can allow a PSTN call to ring/call the phone                                   connected to the FXS port.  
        - It also serves as a life line in case of power                                   outage.  
        What it CANNOT do: 
        - Terminate a VoIP call into the PSTN port 
        - Allow a call from PSTN to route other VoIP devices                                   (different from the FXS phone) over the IP network 
        - Automatically route calls made by the local                                   user to PSTN line 
      Note: On the HT-486 Rev 1.0, the PSTN                                   port is only a life line port that switches to                                   PTSN only on loss of power.
      FXO port: 
        It can support all the functions of a PSTN pass                                   through plus: 
        - Terminate a VoIP call into the PSTN port 
        - Allow a PSTN call to call either the FXS phone                                   or other VoIP devices over the IP network 
        - Route call automatically and transparently to                                   PSTN line according to user configuration 
      Q. Can I call from FXS1 port                                   to FXS2 port? 
      Answer:-
      Yes, they can communicate with each other by                                   dialing the respective extension number. Both                                   FXS ports need a valid sip account registered                                   on the server. 
      Q. How can I disable hook/flash?
      Answer:-
      On the Advanced Settings page, there is a field                                   On-Hook Threshold. One on the selections for this                                   field is &lsquo;Hook/Flash OFF&rsquo;, this option will disable                                   hook/flash on the Phone connected to the ATA.                                   Now, in order to switch to a second channel you                                   will need to press FLASH button on the Phone,                                   instead of doing hook/flash.
      Note:- This feature is not available on older                                   hardware revision models.
      Q. What hardware revision is                                   my Handytone?
      Answer:-
      At the bottom of the device, there will be a                                   white sticker. On this sticker will be a note:
      &nbsp;&lsquo;Rev:                                   x.0&rsquo; where x is the Hardware Revision
      You can also get this information under the Status                                   tab of the Web Configuration pages.
      Your hardware revision is very important information.                                   Depending on your hardware revision, it may/may                                   not be able to upgrade to the latest release. 

      Ex. HT286 Rev 2.0 upgrades upto 1.0.7.19 firmware,                                   while Rev 3.0 can upgrade to the latest release.
      Note:- Once there is a new hardware revision                                   out in the market, the older revision is no more                                   manufactured or sold. 
      Q. How do I access the Web                                   Configuration pages (for HT486/496/488)?
      Answer:-
      Please disconnect all connections for HT486/HT496/HT488                                   and follow the instructions below:-
      1. Connect the Regular Analog Touch tone phone                                   to the HT.
      2. Connect Power Supply.
      3. Now, connect the Ethernet cable between the                                   INTERNET Source (ex. Router, Modem) and the WAN                                   port on the HT.
      4. Now, connect another Ethernet cable between                                   your PC and the LAN port on the HT
      5. Wait for 30 seconds till your PC gets an IP                                   ADDRESS (192.168.2.2)
      6. Now, open Internet Explorer and type in 192.168.2.1,                                   you should see Grandstream Login Screen pop up.
      7. Enter &lsquo;admin&rsquo; as the password
      8. Go to Advanced Settings page and switch &quot;Enable                                   WAN port HTTP access&quot; to YES, hit &lsquo;Update&rsquo;                                   and then &lsquo;Reboot&rsquo;.
      9. Now, disconnect your PC from the LAN port                                   and connect it to any other port on your Router                                   within the same LAN Segment
      10. Now type in the actual IP ADDRESS of the                                   HT (You can look this up by pressing *** on the                                   phone, and then 02) on Internet browser, access                                   the Web Configuration page as you did earlier                                   and configure the device by filling in the information                                   given by your Internet Telephony Service Provider.
      Q. How do I perform a 3-way                                   Conference call?
      Answer:-
      3 way calling between HT, A and B:-
      1. HT calls A
      2. HT talks to A
      3. HT presses FLASH or hook/flash and gets a                                   new dialtone
      4. A is on Hold
      5. HT dials *23 and number for B
      6. HT talks to B
      7. HT presses FLASH or hook/flash to initiate                                   the 3 way calling
      Q. How do I perform Attend                                   Transfer?
      Answer:-
      Attend Transfer from A to B through HT:-
      1. A calls HT
      2. HT talks to A
      3. HT presses FLASH or does hook/flash to get                                   new dialtone.
      4. A is on Hold
      5. HT calls B
      6. HT talks to B
      7. HT hangs up to perform the Attend Transfer.
      8. A and B are in call now.
      Q. Which Phone will ring if                                   there is an incoming call on my Fixed Line (for                                   HT386)?
      Answer:-
      The Phone connected to FXS1 port will ring when                                   there is an incoming call on the Fixed Line.&nbsp; 
      Q. How do I make or receive                                   PSTN calls on my HT (for HT486/386/488)?
      Answer:-
      For receiving a call, the Phone connected to                                   the HT will simply ring.
      For making a call, you will need to press the                                   &lsquo;PSTN Line Access Code&rsquo; first, as configured on                                   the Web Configuration Page (by default it is *00),                                   and then dial the desired PSTN number.
       
      
    
  

]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Grandstream HandyTone FAQ]]></subject>
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<contents><![CDATA[<h3>Crafting a Business VoIP Solution</h3>
Here are some questions that need to be asked when you are considering implementing a Business Voip Solution
<br />
1.) How many offices are you trying to connect - where are they located?<br />
<br />
2.) What type of phone lines do you have at each location?<br />
<br />
3.) What type of internet connection do you have at each location?<br />
<br />
4.) How many phones are at each location?<br />
<br />
5.) What existing network / phone equipment do you have?<br />
<br />
6.) What is your ultimate goal? i.e. to have more features to increase efficiency and effectiveness of your employees by upgrading you phone system? or just to save money and get it done with the least budget as possible?<br />
<br />
7.) How technical are you, and do you have technical people to utilize?<br />
<br />
8.) How much are you spending in total on your monthly reoccurring phone bills (in order to determine what your yearly savings would be for various methods of connecting your offices)<br />
<br />
9.) What does your calling patterns look like? i.e. Do you call outbound internationally to clients all the time from many locations, etc or is the makeup of the calling mostly local in nature?<br />
<br />
]]></contents>
<contentstext><![CDATA[Crafting a Business VoIP Solution
Here are some questions that need to be asked when you are considering implementing a Business Voip Solution
 
1.) How many offices are you trying to connect - where are they located? 
 
2.) What type of phone lines do you have at each location? 
 
3.) What type of internet connection do you have at each location? 
 
4.) How many phones are at each location? 
 
5.) What existing network / phone equipment do you have? 
 
6.) What is your ultimate goal? i.e. to have more features to increase efficiency and effectiveness of your employees by upgrading you phone system? or just to save money and get it done with the least budget as possible? 
 
7.) How technical are you, and do you have technical people to utilize? 
 
8.) How much are you spending in total on your monthly reoccurring phone bills (in order to determine what your yearly savings would be for various methods of connecting your offices) 
 
9.) What does your calling patterns look like? i.e. Do you call outbound internationally to clients all the time from many locations, etc or is the makeup of the calling mostly local in nature? 
 
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Crafting a Business VoIP Solution]]></subject>
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<contents><![CDATA[In most asterisk server deployments, Polycom phones are provisioned via files that are located in the /tftpboot directory of your server.  Every time you reboot a given phone it will pull down its configuration files.  So to make changes to a phone, or a group of phones, you would simple make edits to the files (they are plain text files) and reboot the phone.  The Polycom directory configuration files use the following naming convention: macaddress-directory.xml (it is a regular xml based file).<br />
<br />
For example, if your phone has a MAC address of 0004f215e63c.  It's directory configuration file is called:<br />
<br />
0004f215e63c-directory.xml<br />
<br />
You would simply need to download this file to your computer, make the edits, and then re-upload it to the pbx, overwriting the old file.  Then reboot your phone to effect the change.<br />
<br />
Most linux servers run a "SecureFTP" daemon process which communicates via the SSH protocol (port 22).  This replaces the old standard insecure FTP protocol.  Most modern day Windows FTP client software programs can communicate over the SSH protocol (including WSFTP Pro). <br />
<br />
You can find a detailed explanation of the format of the mac-directory.xml file in the Polycom SIP 2.0 Administrators guide (you can download this on our support site). <br />
<br />
<a href="http://www.888voipstore.com/support/index.php?_m=downloads&_a=viewdownload&downloaditemid=9&nav=0,4" target="_blank">http://www.888voipstore.com/support/index.php?_m=downloads&_a=viewdownload&downloaditemid=9&nav=0,4</a><br />
]]></contents>
<contentstext><![CDATA[In most asterisk server deployments, Polycom phones are provisioned via files that are located in the /tftpboot directory of your server.  Every time you reboot a given phone it will pull down its configuration files.  So to make changes to a phone, or a group of phones, you would simple make edits to the files (they are plain text files) and reboot the phone.  The Polycom directory configuration files use the following naming convention: macaddress-directory.xml (it is a regular xml based file). 
 
For example, if your phone has a MAC address of 0004f215e63c.  It's directory configuration file is called: 
 
0004f215e63c-directory.xml 
 
You would simply need to download this file to your computer, make the edits, and then re-upload it to the pbx, overwriting the old file.  Then reboot your phone to effect the change. 
 
Most linux servers run a "SecureFTP" daemon process which communicates via the SSH protocol (port 22).  This replaces the old standard insecure FTP protocol.  Most modern day Windows FTP client software programs can communicate over the SSH protocol (including WSFTP Pro).  
 
You can find a detailed explanation of the format of the mac-directory.xml file in the Polycom SIP 2.0 Administrators guide (you can download this on our support site).  
 
http://www.888voipstore.com/support/index.php?_m=downloads&_a=viewdownload&downloaditemid=9&nav=0,4 
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[2]]></creatorid>
<author><![CDATA[John Molino]]></author>
<email><![CDATA[mike@voipstore.com]]></email>
<subject><![CDATA[Polycom mac-directory.xml file editing]]></subject>
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<contents><![CDATA[1)	Power down your Polycom Telephone System.<br />
2)	Download newest Polycom Firmware and Launch Pumpkin TFTP Program<br />
2a.	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />
2b.	To download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />
2c.	Browse to the Polycom section and download the Firmware Update for your model of IP telephone.  Save and Unzip the File to your local firmware directory<br />
2d.	Run the Pumpkin Program from its installed directory.<br />
2e.	Once the Program is open. Click the Options button.  This will open an option tab.<br />
2f.	Set the TFTP File system Root to your Firmware directory.  Select the â€œGive filesâ€ and â€œtake filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure   the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. <br />
3)	Power on your Polycom Telephone System.<br />
4)	As the unit initializes, you will be given the option to Start, go to Setup, or Exit.  Chose the setup option on the telephone.<br />
a.	When prompted enter your administrator password.  The default Polycom Password is 456.<br />
5)	You will then be taken to a setup menu.  Scroll down and select the SERVER MENU option.  <br />
6)	Once inside the SERVER MENU, select the SERVER TYPE option and change it to TRIVIAL FTP.  Save your choice and scroll down to the SERVER ADDERESS option.  Enter the IP of the Computer running your Pumpkin Server.  <br />
7)	To save your changes, continue to select exit until you are prompted to save or discard your changes.  Select the SAVE AND REBOOT option.<br />
8)	Your Polycom Telephone will reboot and begin to update. This will take as long as fifteen to thirty minutes.  <br />
9)	Congratulations, you have completed the loading of firmware on your Polycom Telephone system!<br />
]]></contents>
<contentstext><![CDATA[1)	Power down your Polycom Telephone System.
2)	Download newest Polycom Firmware and Launch Pumpkin TFTP Program
2a.	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)
2b.	To download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.
2c.	Browse to the Polycom section and download the Firmware Update for your model of IP telephone.  Save and Unzip the File to your local firmware directory
2d.	Run the Pumpkin Program from its installed directory.
2e.	Once the Program is open. Click the Options button.  This will open an option tab.
2f.	Set the TFTP File system Root to your Firmware directory.  Select the â€œGive filesâ€ and â€œtake filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure   the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. 
3)	Power on your Polycom Telephone System.
4)	As the unit initializes, you will be given the option to Start, go to Setup, or Exit.  Chose the setup option on the telephone.
a.	When prompted enter your administrator password.  The default Polycom Password is 456.
5)	You will then be taken to a setup menu.  Scroll down and select the SERVER MENU option.  
6)	Once inside the SERVER MENU, select the SERVER TYPE option and change it to TRIVIAL FTP.  Save your choice and scroll down to the SERVER ADDERESS option.  Enter the IP of the Computer running your Pumpkin Server.  
7)	To save your changes, continue to select exit until you are prompted to save or discard your changes.  Select the SAVE AND REBOOT option.
8)	Your Polycom Telephone will reboot and begin to update. This will take as long as fifteen to thirty minutes.  
9)	Congratulations, you have completed the loading of firmware on your Polycom Telephone system!
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Install Firmware on a Polycom Phone]]></subject>
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<contents><![CDATA[1)	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />
2)	To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />
2a.	Browse to the Aastra section and download the Firmware Update for your model of IP telephone.<br />
2b.	Save and Unzip the File to your local directory<br />
3)	Run the Pumpkin program from its install directory.  <br />
3a.	Once the Program is open, Click the options button.  This will open an option tab.<br />
3b.	Set the TFTP file system to your Firmware directory.  Select the â€œGive Filesâ€ and â€œTake Filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. <br />
4)	Find your Aastra Telephoneâ€™s IP address.<br />
4a.	To find the address, enter the phoneâ€™s Options menu.    Scroll down and select the Network option.<br />
4b.	When prompted, enter your password. The Default setting for Aastra phones is Password: 22222.<br />
4c.	Scroll down the network menu to the IP Address option and select it.  This will display your telephoneâ€™s IP address. <br />
5)	Open an internet browser.  Enter your telephoneâ€™s IP address into the navigation bar. <br />
5a.	When prompted, enter your username and password. The Default setting for Aastra phones is Username: Admin, Password: 22222. <br />
6)	This will display your Telephones web interface.<br />
7)	Select the FIRMWARE UPDATE link on the bottom left side of the screen.<br />
8)	This will display the Firmware Update interface.  <br />
8a.	In the TFTP Server IP box enter the IP of your Pumpkin Server.<br />
8b.	In the File Name box enter the file name of the firmware file that is to be loaded (See below image for an example.) <br />
8c.	Click Download Firmware.  This will begin the firmware update, which will take several minutes.  Once complete your IP telephone will reboot.  Once the telephone returns to its normal user interface, the update is complete. <br />
 <br />
<br />
<br />
<br />
<br />
]]></contents>
<contentstext><![CDATA[1)	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)
2)	To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.
2a.	Browse to the Aastra section and download the Firmware Update for your model of IP telephone.
2b.	Save and Unzip the File to your local directory
3)	Run the Pumpkin program from its install directory.  
3a.	Once the Program is open, Click the options button.  This will open an option tab.
3b.	Set the TFTP file system to your Firmware directory.  Select the â€œGive Filesâ€ and â€œTake Filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. 
4)	Find your Aastra Telephoneâ€™s IP address.
4a.	To find the address, enter the phoneâ€™s Options menu.    Scroll down and select the Network option.
4b.	When prompted, enter your password. The Default setting for Aastra phones is Password: 22222.
4c.	Scroll down the network menu to the IP Address option and select it.  This will display your telephoneâ€™s IP address. 
5)	Open an internet browser.  Enter your telephoneâ€™s IP address into the navigation bar. 
5a.	When prompted, enter your username and password. The Default setting for Aastra phones is Username: Admin, Password: 22222. 
6)	This will display your Telephones web interface.
7)	Select the FIRMWARE UPDATE link on the bottom left side of the screen.
8)	This will display the Firmware Update interface.  
8a.	In the TFTP Server IP box enter the IP of your Pumpkin Server.
8b.	In the File Name box enter the file name of the firmware file that is to be loaded (See below image for an example.) 
8c.	Click Download Firmware.  This will begin the firmware update, which will take several minutes.  Once complete your IP telephone will reboot.  Once the telephone returns to its normal user interface, the update is complete. 
 




]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Update Firmware on an Aastra Phone]]></subject>
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<contents><![CDATA[1)	Power down your Aastra Telephone System.<br />
2)	Hold down the number 1 and # key(s). <br />
3)	Power on your Aastra Telephone System.<br />
4)	Continue to hold down 1 and # key(s) while the system initializes.  <br />
5)	Once the system initializes, you will receive the following message:<br />
â€œApplication Not Foundâ€<br />
â€œTo restore go to Web Recovery Pageâ€<br />
â€œWeb recovery page can be found at the following IP Address: XXX.XXX.XXX.XXX<br />
6)	Enter IP Address into your web browser to visit the Web Recovery Interface.<br />
6a.	The Web Recovery will ask for 2 fields to be filled: File Name and TFTP Server.  File Name will be the name of your downloaded Firmware, and the TFTP Server Will be the IP of a PC running a TFTP program like Pumpkin.<br />
6b.	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />
6c.	To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />
6d.	Browse to the Aastra section and download the Firmware Update for your model of IP telephone.<br />
6e.	Save and Unzip the File to your local directory<br />
6f.	Run the Pumpkin Program from its installed directory.<br />
6g.	Once the Program is open. Click the Options button.  This will open an option tab.<br />
6h.	Set the TFTP File system Root to your Firmware directory.  Select the â€œGive filesâ€ and â€œtake filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure the Pumpkin Server is running by verifying in the bottom right corner of  the Pumpkin window. <br />
6i.	Return to your internet browser, which should still be at the Web Recovery Interface.   Enter the exact file name of the firmware to be installed in the â€œfilenameâ€ box.  Enter the IP address of the Computer running Pumpkin into the TFTP box.   <br />
6j.	Select the download firmware button.  Install will take around 30 seconds.  <br />
7)	You have completed the Firmware Installation of an Aastra IP Telephone.<br />
]]></contents>
<contentstext><![CDATA[1)	Power down your Aastra Telephone System.
2)	Hold down the number 1 and # key(s). 
3)	Power on your Aastra Telephone System.
4)	Continue to hold down 1 and # key(s) while the system initializes.  
5)	Once the system initializes, you will receive the following message:
â€œApplication Not Foundâ€
â€œTo restore go to Web Recovery Pageâ€
â€œWeb recovery page can be found at the following IP Address: XXX.XXX.XXX.XXX
6)	Enter IP Address into your web browser to visit the Web Recovery Interface.
6a.	The Web Recovery will ask for 2 fields to be filled: File Name and TFTP Server.  File Name will be the name of your downloaded Firmware, and the TFTP Server Will be the IP of a PC running a TFTP program like Pumpkin.
6b.	Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)
6c.	To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.
6d.	Browse to the Aastra section and download the Firmware Update for your model of IP telephone.
6e.	Save and Unzip the File to your local directory
6f.	Run the Pumpkin Program from its installed directory.
6g.	Once the Program is open. Click the Options button.  This will open an option tab.
6h.	Set the TFTP File system Root to your Firmware directory.  Select the â€œGive filesâ€ and â€œtake filesâ€ options.  Apply the settings and select â€œOKâ€ to return to Pumpkin.  Ensure the Pumpkin Server is running by verifying in the bottom right corner of  the Pumpkin window. 
6i.	Return to your internet browser, which should still be at the Web Recovery Interface.   Enter the exact file name of the firmware to be installed in the â€œfilenameâ€ box.  Enter the IP address of the Computer running Pumpkin into the TFTP box.   
6j.	Select the download firmware button.  Install will take around 30 seconds.  
7)	You have completed the Firmware Installation of an Aastra IP Telephone.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Install Firmware on an Aastra Phone]]></subject>
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<contents><![CDATA[VoIP : Voice over Internet Protocol / Voice over IP<br />
ADSL : Asymmetric digital subscriber line. Type of DSL connection where data flow is greater in one direction than the other.<br />
ATA : Analog Telephone Adapter<br />
Unit to connect a regular anaog phone to the internet to make and receive VoIP calls. Meaning you can use your existing analog phone (eg Uniden cordless, Telstra corded) and start making VoIP calls. <br />
Automated Attendent : Automated receptionist answering calls using a recorded message and directing calls to the required area of the business. <br />
Bandwidth : The transmission capacity of a specific device or network.<br />
Bridge : A device that connects two different king od local networks. ie wireless network to a wired ethernet network.<br />
Broadband : An always on, fast internet connection. Recommended for VoIP<br />
CID : Caller ID/Caller ID Display<br />
Codec : Coder-decoder<br />
Software that converts audio signals to compressed digital signals.<br />
DHCP : Cynamic Host Configuration Protocol. A protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers.<br />
DMZ : Demilitarized Zone : Removes the router's firewall protection from on PC, allowing it to be "seen" from the internet.<br />
DID : Direct in Dial - allows callers to call a specific extension/person and bypass the main reception number/automated attendent.<br />
DSL : Digital Subscriber Line. An always on broadband connection over traditional phone lines. <br />
Firmware : The embedded software that runs a networking device.<br />
Full Duplex : The ability of a networking device to receive and transmit data simultaneously.<br />
Gateway : A device that interconnects networks with different, incompatible communications protocols.<br />
IP : Internet Protocol.<br />
IP Address : The address used to identify a computer or device on a network. (example 192.0.00.0)<br />
IPSec : Internet Protocol Security. A VPN protocol used to implement secure exchange of packets at the IP layer. Most widely used for enabling virtual private networks.<br />
ITSP : Internet Telephony Service Provider. A company that provides voice communicationg service over the Internet (VoIP provider) ie engin, gotalk, MyNetFone, etc.<br />
IVR : Interactive Voice Response<br />
LAN : Local Area Network. The computers and networking products that make up the network in your home of office.<br />
MAC Address : Media Access Control. The unique address that a manufacturer assigns to each networking device. Typically 12 digits in length.<br />
Managed Switch : A network switch with an IP address that lets you monitor and administer your network.<br />
NAT : Network Address Translation<br />
PBX : Private Branch Exchange : A private telephone network used within a business. Users share a certain number of outside lines to make external phone calls. See Linksys SPA9000, epygi Quadro, Switchvox, Asterisk<br />
PoE : Power over Ethernet : A technology enabling and Ethernet network cable to deliver both data and power.<br />
POTS : Plain old telephone service<br />
PSTN : Public Switch Telephone Network<br />
QoS : Quality of Service<br />
SIP : Session Initiation Protocol<br />
Softphone : Telephony Software  installed on a PC/Mac enabling the user to make and receive phone calls over the internet connection. Requires software, PC, microphone and headset/speakers.<br />
]]></contents>
<contentstext><![CDATA[VoIP : Voice over Internet Protocol / Voice over IP
ADSL : Asymmetric digital subscriber line. Type of DSL connection where data flow is greater in one direction than the other.
ATA : Analog Telephone Adapter
Unit to connect a regular anaog phone to the internet to make and receive VoIP calls. Meaning you can use your existing analog phone (eg Uniden cordless, Telstra corded) and start making VoIP calls. 
Automated Attendent : Automated receptionist answering calls using a recorded message and directing calls to the required area of the business. 
Bandwidth : The transmission capacity of a specific device or network.
Bridge : A device that connects two different king od local networks. ie wireless network to a wired ethernet network.
Broadband : An always on, fast internet connection. Recommended for VoIP
CID : Caller ID/Caller ID Display
Codec : Coder-decoder
Software that converts audio signals to compressed digital signals.
DHCP : Cynamic Host Configuration Protocol. A protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers.
DMZ : Demilitarized Zone : Removes the router's firewall protection from on PC, allowing it to be "seen" from the internet.
DID : Direct in Dial - allows callers to call a specific extension/person and bypass the main reception number/automated attendent.
DSL : Digital Subscriber Line. An always on broadband connection over traditional phone lines. 
Firmware : The embedded software that runs a networking device.
Full Duplex : The ability of a networking device to receive and transmit data simultaneously.
Gateway : A device that interconnects networks with different, incompatible communications protocols.
IP : Internet Protocol.
IP Address : The address used to identify a computer or device on a network. (example 192.0.00.0)
IPSec : Internet Protocol Security. A VPN protocol used to implement secure exchange of packets at the IP layer. Most widely used for enabling virtual private networks.
ITSP : Internet Telephony Service Provider. A company that provides voice communicationg service over the Internet (VoIP provider) ie engin, gotalk, MyNetFone, etc.
IVR : Interactive Voice Response
LAN : Local Area Network. The computers and networking products that make up the network in your home of office.
MAC Address : Media Access Control. The unique address that a manufacturer assigns to each networking device. Typically 12 digits in length.
Managed Switch : A network switch with an IP address that lets you monitor and administer your network.
NAT : Network Address Translation
PBX : Private Branch Exchange : A private telephone network used within a business. Users share a certain number of outside lines to make external phone calls. See Linksys SPA9000, epygi Quadro, Switchvox, Asterisk
PoE : Power over Ethernet : A technology enabling and Ethernet network cable to deliver both data and power.
POTS : Plain old telephone service
PSTN : Public Switch Telephone Network
QoS : Quality of Service
SIP : Session Initiation Protocol
Softphone : Telephony Software  installed on a PC/Mac enabling the user to make and receive phone calls over the internet connection. Requires software, PC, microphone and headset/speakers.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[VoIP Glossary : What do the acronyms and VoIP terms mean?]]></subject>
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<contents><![CDATA[1)	Add desired extension to your PBX system.  Ensure you set a proper Authorization ID and Password.<br />
2)	Ensure that your system is wired correctly (LAN, Phone and power cables.) <br />
3)	Power on system.  Wait several seconds while system initializes. <br />
4)	Once system has finished start up, enter your phone system menu.  Scroll down to and select NETWORK.<br />
5)	Once inside the NETWORK menu, scroll to and select IP ADDRESS.  This will display your Unitâ€™s current IP address.<br />
6)	Open a Web Browser program.  Enter your Telephoneâ€™s IP address into the navigation bar.    This will display the Linksys Web Interface.<br />
7)	Click the ADMIN LOGIN link on the top right hand of the screen.  (NOTE: Linksys default username and password for ADMIN functions are username: admin, password: admin)<br />
8)	Once the ADMIN page opens, select EXT 1 from the taskbar at the top, this will let you configure your phone line.<br />
9)	Once the EXT 1 page displays, enter your information in the following fields:<br />
                <br />
                PROXY AND REGISTRATION<br />
                Proxy: Enter your PBX IP Address<br />
<br />
                SUBSCRIBER INFO<br />
                User ID= User ID that you set up with the new extension.<br />
                Auth ID= Authorization ID that you set up with the new extension.<br />
                Password= Password corresponding to the Auth ID you set up with the new extension.<br />
<br />
                Note: Change the Use Auth ID box from NO to YES.<br />
<br />
10)	Click Submit changes, and wait as your system updates and reboots.  <br />
11)	You have completed a phone registration for your new Linksys IP Telephone System.<br />
]]></contents>
<contentstext><![CDATA[1)	Add desired extension to your PBX system.  Ensure you set a proper Authorization ID and Password.
2)	Ensure that your system is wired correctly (LAN, Phone and power cables.) 
3)	Power on system.  Wait several seconds while system initializes. 
4)	Once system has finished start up, enter your phone system menu.  Scroll down to and select NETWORK.
5)	Once inside the NETWORK menu, scroll to and select IP ADDRESS.  This will display your Unitâ€™s current IP address.
6)	Open a Web Browser program.  Enter your Telephoneâ€™s IP address into the navigation bar.    This will display the Linksys Web Interface.
7)	Click the ADMIN LOGIN link on the top right hand of the screen.  (NOTE: Linksys default username and password for ADMIN functions are username: admin, password: admin)
8)	Once the ADMIN page opens, select EXT 1 from the taskbar at the top, this will let you configure your phone line.
9)	Once the EXT 1 page displays, enter your information in the following fields:
                
                PROXY AND REGISTRATION
                Proxy: Enter your PBX IP Address

                SUBSCRIBER INFO
                User ID= User ID that you set up with the new extension.
                Auth ID= Authorization ID that you set up with the new extension.
                Password= Password corresponding to the Auth ID you set up with the new extension.

                Note: Change the Use Auth ID box from NO to YES.

10)	Click Submit changes, and wait as your system updates and reboots.  
11)	You have completed a phone registration for your new Linksys IP Telephone System.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Register/Install a Linksys IP Telephone System]]></subject>
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<contents><![CDATA[Here is a complete guide of the Grandstream IVR commands for use with the Handytone Series.<br />
<br />
Menu Voice Prompt Options<br />
<br />
To enter this voice prompt menu, press the LED button or press â€œ***â€ from the analog phone.<br />
<br />
Main Menu â€œEnter a Menu Optionâ€ Press â€œ*â€ for the next menu option<br />
Press â€œ#â€ to return to the main menu<br />
<br />
Available Menu Options: 01-06, 47, 86, 99.<br />
<br />
01 â€œDHCP Modeâ€, â€œStatic IP Modeâ€<br />
Press â€œ9â€ to toggle the selection<br />
<br />
If using â€œStatic IP Modeâ€, configure the IP address information using menus 02 to 05.<br />
If using â€œDynamic IP Modeâ€, all IP address information comes from the DHCP server automatically after reboot.<br />
<br />
02 â€œIP Address â€œ + IP address The current WAN IP address is announced<br />
If using â€œStatic IP Modeâ€, enter 12 digit new IP address.<br />
<br />
03 â€œSubnet â€œ + IP address Same as menu 02<br />
<br />
04 â€œGateway â€œ + IP address Same as menu 02<br />
<br />
05 â€œDNS Server â€œ + IP address Same as menu 02<br />
<br />
07 Preferred Vocoder Press â€œ9â€ to move to the next selection in the list:<br />
â€¢ PCM U / PCM A<br />
â€¢ G.723<br />
â€¢ G.729<br />
â€¢ G.726<br />
â€¢ iLBC<br />
<br />
12 WAN Port Web Access Press â€œ9â€ to toggle between enable / disable<br />
<br />
13 Firmware Server IP Address<br />
Announces current Firmware Server IP address. Enter 12 digit new IP address.<br />
<br />
14 Configuration Server IP Address Announces current Config Server Path IP address.<br />
Enter 12 digit new IP address.<br />
<br />
15 Upgrade Protocol Upgrade protocol for firmware and configuration update.<br />
Press â€œ9â€ to toggle between TFTP / HTTP<br />
<br />
16 Firmware Version Firmware version information.<br />
<br />
17 Firmware Upgrade Firmware upgrade mode. Press â€œ9â€ to toggle among the<br />
following three options:<br />
- always check<br />
- check when pre/suffix changes<br />
- never upgrade<br />
<br />
47 â€œDirect IP Callingâ€ Enter a 12 digit IP address to make a direct IP call, after dial tone. (See â€œMake a Direct IP Callâ€.)<br />
<br />
99 â€œRESETâ€ Press â€œ9â€ to reboot the device; or<br />
Enter encoded MAC address to restore factory default<br />
setting.<br />
<br />
RESET VIA IVR<br />
1.) Find the MAC address of the device. It is a 12 digits HEX number located on the bottom<br />
of the unit.<br />
<br />
2.) Encode the MAC address. Please use the following mapping:<br />
0-9: 0-9<br />
A: 22<br />
B: 222<br />
C: 2222<br />
D: 33<br />
E: 333<br />
F: 3333<br />
For example, if the MAC address is 000b8200e395, it should be encoded as<br />
â€œ0002228200333395â€.<br />
<br />
3.) To perform factory reset:<br />
a.) Pick up the headset and dial â€œ***â€ for voice prompt.<br />
b.) Enter â€œ99â€ and get the voice prompt â€œResetâ€.<br />
c.) Enter the encoded MAC address of the device.<br />
d.) Wait for 15 seconds. The device will reboot automatically and restore to factory<br />
default setting.<br />
<br />
Any â€œInvalid Entryâ€ Automatically returns to main menu.<br />
<br />
NOTE:<br />
â€¢ Once the button is pressed, you will hear the voice prompt main menu. If the button is pressed<br />
again, while it is already in the voice prompt menu, it jumps to â€œDirect IP Callâ€ option and a dial<br />
tone is prompted<br />
â€¢ â€œ*â€ shifts down to the next menu option<br />
â€¢ â€œ#â€ returns to the main menu<br />
â€¢ â€œ9â€ functions as the ENTER key in many cases to confirm an option<br />
â€¢ All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP<br />
address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26<br />
should be key in like 192168000026, no dot needed while input). Once all of the digits are<br />
collected, the input will be processed.<br />
â€¢ Key entry can not be deleted but the phone may prompt error once it is detected<br />
]]></contents>
<contentstext><![CDATA[Here is a complete guide of the Grandstream IVR commands for use with the Handytone Series.

Menu Voice Prompt Options

To enter this voice prompt menu, press the LED button or press â€œ***â€ from the analog phone.

Main Menu â€œEnter a Menu Optionâ€ Press â€œ*â€ for the next menu option
Press â€œ#â€ to return to the main menu

Available Menu Options: 01-06, 47, 86, 99.

01 â€œDHCP Modeâ€, â€œStatic IP Modeâ€
Press â€œ9â€ to toggle the selection

If using â€œStatic IP Modeâ€, configure the IP address information using menus 02 to 05.
If using â€œDynamic IP Modeâ€, all IP address information comes from the DHCP server automatically after reboot.

02 â€œIP Address â€œ + IP address The current WAN IP address is announced
If using â€œStatic IP Modeâ€, enter 12 digit new IP address.

03 â€œSubnet â€œ + IP address Same as menu 02

04 â€œGateway â€œ + IP address Same as menu 02

05 â€œDNS Server â€œ + IP address Same as menu 02

07 Preferred Vocoder Press â€œ9â€ to move to the next selection in the list:
â€¢ PCM U / PCM A
â€¢ G.723
â€¢ G.729
â€¢ G.726
â€¢ iLBC

12 WAN Port Web Access Press â€œ9â€ to toggle between enable / disable

13 Firmware Server IP Address
Announces current Firmware Server IP address. Enter 12 digit new IP address.

14 Configuration Server IP Address Announces current Config Server Path IP address.
Enter 12 digit new IP address.

15 Upgrade Protocol Upgrade protocol for firmware and configuration update.
Press â€œ9â€ to toggle between TFTP / HTTP

16 Firmware Version Firmware version information.

17 Firmware Upgrade Firmware upgrade mode. Press â€œ9â€ to toggle among the
following three options:
- always check
- check when pre/suffix changes
- never upgrade

47 â€œDirect IP Callingâ€ Enter a 12 digit IP address to make a direct IP call, after dial tone. (See â€œMake a Direct IP Callâ€.)

99 â€œRESETâ€ Press â€œ9â€ to reboot the device; or
Enter encoded MAC address to restore factory default
setting.

RESET VIA IVR
1.) Find the MAC address of the device. It is a 12 digits HEX number located on the bottom
of the unit.

2.) Encode the MAC address. Please use the following mapping:
0-9: 0-9
A: 22
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, if the MAC address is 000b8200e395, it should be encoded as
â€œ0002228200333395â€.

3.) To perform factory reset:
a.) Pick up the headset and dial â€œ***â€ for voice prompt.
b.) Enter â€œ99â€ and get the voice prompt â€œResetâ€.
c.) Enter the encoded MAC address of the device.
d.) Wait for 15 seconds. The device will reboot automatically and restore to factory
default setting.

Any â€œInvalid Entryâ€ Automatically returns to main menu.

NOTE:
â€¢ Once the button is pressed, you will hear the voice prompt main menu. If the button is pressed
again, while it is already in the voice prompt menu, it jumps to â€œDirect IP Callâ€ option and a dial
tone is prompted
â€¢ â€œ*â€ shifts down to the next menu option
â€¢ â€œ#â€ returns to the main menu
â€¢ â€œ9â€ functions as the ENTER key in many cases to confirm an option
â€¢ All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP
address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26
should be key in like 192168000026, no dot needed while input). Once all of the digits are
collected, the input will be processed.
â€¢ Key entry can not be deleted but the phone may prompt error once it is detected
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[Grandstream Handytone Interactive Voice Response Menu]]></subject>
<isedited><![CDATA[1]]></isedited>
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<contents><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to the email sent to the registered user on the CogentVoip account with the title â€˜CogentVoip Account Setupâ€™. This email is sent after you register and contains the account information you need to complete the setup process. From within the 3CX configuration screens:<br />
1. Under Lines, click â€œAdd VOIPâ€<br />
2. Select the radio button labeled select â€œGeneric Voip Providerâ€ enter â€œCogentVoipâ€ as the Name of Provider and click â€œNextâ€.<br />
3. Enter â€˜CogentVoipâ€™ in both the Registrar and Outbound Proxy fields.  Click on the â€˜Source Identificationâ€™ tab, then under â€˜SIP Fieldâ€™, select â€˜From: Host Partâ€™.  For the Value drop down, select â€˜Custom Fieldâ€™.  Set the Custom value field to â€˜CogentVoip.comâ€™, then select the match any fields option.  Click â€œNextâ€ again.<br />
4. In the External Number field, enter a 1 followed by your DID (11 total digits).<br />
5. From the CogentVoip Account Setup email, obtain the â€˜Authentication Usernameâ€™ and enter it in to the â€˜Authentication IDâ€™ field of the 3CX. Please ensure you have no hidden characters or trailing spaces. <br />
6. From the CogentVoip Account Setup email, obtain the â€˜Authentication Passwordâ€™ and enter it in to the â€˜Authentication Passwordâ€™ field of the 3CX. Please ensure you have no hidden characters or trailing spaces. Click â€œNextâ€.<br />
7. Enter any Outbound Rules you wish to impose on your network. Click â€œFinishâ€.<br />
8. Your Account should now be registered; you can view the status of the Voip Trunk by clicking â€˜Line Statusâ€™, which is located under â€˜Phone Systemâ€™  on the left hand side of the 3CX Web Interface.<br />
]]></contents>
<contentstext><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to the email sent to the registered user on the CogentVoip account with the title â€˜CogentVoip Account Setupâ€™. This email is sent after you register and contains the account information you need to complete the setup process. From within the 3CX configuration screens:
1. Under Lines, click â€œAdd VOIPâ€
2. Select the radio button labeled select â€œGeneric Voip Providerâ€ enter â€œCogentVoipâ€ as the Name of Provider and click â€œNextâ€.
3. Enter â€˜CogentVoipâ€™ in both the Registrar and Outbound Proxy fields.  Click on the â€˜Source Identificationâ€™ tab, then under â€˜SIP Fieldâ€™, select â€˜From: Host Partâ€™.  For the Value drop down, select â€˜Custom Fieldâ€™.  Set the Custom value field to â€˜CogentVoip.comâ€™, then select the match any fields option.  Click â€œNextâ€ again.
4. In the External Number field, enter a 1 followed by your DID (11 total digits).
5. From the CogentVoip Account Setup email, obtain the â€˜Authentication Usernameâ€™ and enter it in to the â€˜Authentication IDâ€™ field of the 3CX. Please ensure you have no hidden characters or trailing spaces. 
6. From the CogentVoip Account Setup email, obtain the â€˜Authentication Passwordâ€™ and enter it in to the â€˜Authentication Passwordâ€™ field of the 3CX. Please ensure you have no hidden characters or trailing spaces. Click â€œNextâ€.
7. Enter any Outbound Rules you wish to impose on your network. Click â€œFinishâ€.
8. Your Account should now be registered; you can view the status of the Voip Trunk by clicking â€˜Line Statusâ€™, which is located under â€˜Phone Systemâ€™  on the left hand side of the 3CX Web Interface.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to configure 3CX with CogentVoip Service]]></subject>
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<contents><![CDATA[Step 1 â€“ NAT Settings<br />
----------------------------<br />
The first stage is to ensure that the NAT (Network Address Translation) configuration is<br />
correct in your sip_nat.conf.<br />
<br />
Login as maint and the password you have set, find the Asterisk link and hover over it.  You should see â€œConfig Editâ€. Select and look for â€œsip_nat.confâ€, enter the following lines in iof which are in quotes;<br />
<br />
â€œnat=yesâ€ <br />
-This tells the Asterisk box that some adjustments will need to be made while using SIP protocol over a WAN link.<br />
<br />
For fixed IP addresses:<br />
<br />
â€œexternip=XXX.XXX.XXX.XXX â€œ<br />
-This sets the IP address in the SIP packets to your outside IP address. Use the external IP address you have from your ISP.<br />
<br />
â€œlocalnet=192.168.X.X/255.255.255.0â€<br />
 -This enables the trixbox to know who is registered from the inside LAN and allows the SIP messages to be generated with the correct IP addresses. Use your local subnet settings.<br />
<br />
Also, under the Security Settings tab, set "Allow Anonymous SIP Calls" to "Yes".<br />
<br />
Step 2 â€“ Trunk Settings<br />
----------------------------<br />
Trunk Name : cogentvoip.com<br />
Peer details:<br />
dtmf=rfc2833<br />
dtmfmode=rfc2833<br />
fromuser= local user name [can be used for CLI generation at user level]<br />
host=cogentvoip.com<br />
insecure=very<br />
nat=yes<br />
qualify=no<br />
secret= [CogentVoip issued proxy password. This is not the web login password]<br />
type=peer<br />
username=[CogentVoip issued proxy user name. This is not the web login user name]<br />
<br />
<br />
User Context=[local user name]<br />
User Details<br />
context=from-trunk<br />
fromuser= local user name [can be used for CLI generation at user level]<br />
host=your External IP address<br />
insecure=very<br />
nat=yes<br />
qualify=no<br />
secret=[CogentVoip issued proxy password. This is not the web login password]<br />
type=user<br />
username=[CogentVoip issued proxy user name. This is not the web login user name]<br />
<br />
<br />
Register String: Username:Password@cogentvoip.com<br />
<br />
Create your inbound routes for inbound numbers, e.g. 7035790200 routes to phone A,<br />
7035790201 routes to phone B and that should cover your CogentVoip trunk setup.<br />
<br />
<br />
Firewall/Router<br />
----------------------------<br />
You will need to open port 5060 on your firewall to enable the SIP signaling to traverse.<br />
You will also need to open the RTP or audio ports. This is different for each customer<br />
premise device. Please reference trixbox for this detail.<br />
]]></contents>
<contentstext><![CDATA[Step 1 â€“ NAT Settings
----------------------------
The first stage is to ensure that the NAT (Network Address Translation) configuration is
correct in your sip_nat.conf.

Login as maint and the password you have set, find the Asterisk link and hover over it.  You should see â€œConfig Editâ€. Select and look for â€œsip_nat.confâ€, enter the following lines in iof which are in quotes;

â€œnat=yesâ€ 
-This tells the Asterisk box that some adjustments will need to be made while using SIP protocol over a WAN link.

For fixed IP addresses:

â€œexternip=XXX.XXX.XXX.XXX â€œ
-This sets the IP address in the SIP packets to your outside IP address. Use the external IP address you have from your ISP.

â€œlocalnet=192.168.X.X/255.255.255.0â€
 -This enables the trixbox to know who is registered from the inside LAN and allows the SIP messages to be generated with the correct IP addresses. Use your local subnet settings.

Also, under the Security Settings tab, set "Allow Anonymous SIP Calls" to "Yes".

Step 2 â€“ Trunk Settings
----------------------------
Trunk Name : cogentvoip.com
Peer details:
dtmf=rfc2833
dtmfmode=rfc2833
fromuser= local user name [can be used for CLI generation at user level]
host=cogentvoip.com
insecure=very
nat=yes
qualify=no
secret= [CogentVoip issued proxy password. This is not the web login password]
type=peer
username=[CogentVoip issued proxy user name. This is not the web login user name]


User Context=[local user name]
User Details
context=from-trunk
fromuser= local user name [can be used for CLI generation at user level]
host=your External IP address
insecure=very
nat=yes
qualify=no
secret=[CogentVoip issued proxy password. This is not the web login password]
type=user
username=[CogentVoip issued proxy user name. This is not the web login user name]


Register String: Username:Password@cogentvoip.com

Create your inbound routes for inbound numbers, e.g. 7035790200 routes to phone A,
7035790201 routes to phone B and that should cover your CogentVoip trunk setup.


Firewall/Router
----------------------------
You will need to open port 5060 on your firewall to enable the SIP signaling to traverse.
You will also need to open the RTP or audio ports. This is different for each customer
premise device. Please reference trixbox for this detail.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to configure trixbox CE  with CogentVoip Service]]></subject>
<isedited><![CDATA[1]]></isedited>
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<contents><![CDATA[Polycom phones support speed dial configuration via the phone's interface.<br />
<br />
<br />
Configure Speed Dial Keys<br />
<br />
   1. Press Menu<br />
   2. Press 1 Features<br />
   3. Press 1 Contact Directory<br />
   4. Press Add<br />
   5. Fill out the settings<br />
         1. Scenario #1 - Speed Dial an Employee<br />
         2. Fill out the Employees First and Last name in the appropriate fields<br />
         3. Type in the Employee's extension in the Contact field<br />
         4. Leave the Speed Dial Index at the default value (increments automatically)<br />
         5. Press Save<br />
         6. The configured speed dial will appear as Line 2 (for the first contact) and increment with each new configured speed dial (Line 3, Line 4, etc)<br />
         7. Scenario #2 - Speed Dial Park/Voicemail/Other System Extension<br />
         8. For First and Last name - enter whatever text will help you easily identify the speed dial function (e.g. if you want to speed dial Park, type in Park and leave the other<br />
             name field blank)<br />
         9. Type in the system extension (Park = 9000, Voicemail = 8555, Group Intercom = *74+Group Extension, etc.).  Remember you can only type in ONE extension per <br />
             speed dial key<br />
        10. Press Save<br />
        11. The configured speed dial will appear as Line 2 (for the first key) and increment with each new configured speed dial<br />
<br />
   6. When you are finished configuring all your speed dial keys, press the Menu button to return to the default screen<br />
<br />
Executing a Speed Dial<br />
<br />
   1. Press the button - the phone will dial the extension you configured.<br />
<br />
Executing a Speed Dial to a System Extension<br />
<br />
Customers commonly wish to speed dial Park.  Once you setup Park as a speed dial on your phone, you can quickly transfer callers to Park via the following sequence:<br />
<br />
   1. Press the softkey Txfr (may be labeled Trnsfer on some versions of Polycom firmware)<br />
   2. Press the Park speed dial key you previously configured<br />
   3. Listen for the Parking extension<br />
   4. Hangup<br />
   5. Inform the destination party of the parked extension<br />
   6. That person should dial that extension (e.g. 9001) from their own phone to reconnect with the caller<br />
]]></contents>
<contentstext><![CDATA[Polycom phones support speed dial configuration via the phone's interface.


Configure Speed Dial Keys

   1. Press Menu
   2. Press 1 Features
   3. Press 1 Contact Directory
   4. Press Add
   5. Fill out the settings
         1. Scenario #1 - Speed Dial an Employee
         2. Fill out the Employees First and Last name in the appropriate fields
         3. Type in the Employee's extension in the Contact field
         4. Leave the Speed Dial Index at the default value (increments automatically)
         5. Press Save
         6. The configured speed dial will appear as Line 2 (for the first contact) and increment with each new configured speed dial (Line 3, Line 4, etc)
         7. Scenario #2 - Speed Dial Park/Voicemail/Other System Extension
         8. For First and Last name - enter whatever text will help you easily identify the speed dial function (e.g. if you want to speed dial Park, type in Park and leave the other
             name field blank)
         9. Type in the system extension (Park = 9000, Voicemail = 8555, Group Intercom = *74+Group Extension, etc.).  Remember you can only type in ONE extension per 
             speed dial key
        10. Press Save
        11. The configured speed dial will appear as Line 2 (for the first key) and increment with each new configured speed dial

   6. When you are finished configuring all your speed dial keys, press the Menu button to return to the default screen

Executing a Speed Dial

   1. Press the button - the phone will dial the extension you configured.

Executing a Speed Dial to a System Extension

Customers commonly wish to speed dial Park.  Once you setup Park as a speed dial on your phone, you can quickly transfer callers to Park via the following sequence:

   1. Press the softkey Txfr (may be labeled Trnsfer on some versions of Polycom firmware)
   2. Press the Park speed dial key you previously configured
   3. Listen for the Parking extension
   4. Hangup
   5. Inform the destination party of the parked extension
   6. That person should dial that extension (e.g. 9001) from their own phone to reconnect with the caller
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Program keys on a Polycom IP Phone]]></subject>
<isedited><![CDATA[1]]></isedited>
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<kbarticle>
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<contents><![CDATA[Phones local to the trixbox Pro use a Network Time Service in order to synchronize with the accurate, local time.  This Time Service comes in the form of an IP address given to the phone by the trixbox Pro (If used as a DHCP server) or by your own DHCP server.  <br />
<br />
This technology is called NTP, and though Fonality tries to automate the process of a new phone booting up and correctly synchronizing the time for your location, when the phone is moved from the office to a remote location (including a remote office that does not have a trixbox Pro installed on-site), the automated process will fail.<br />
<br />
Fonality uses the IP address of your trixbox Pro as the NTP value for every phone.  When the phone is moved to a remote location, it stores this IP address.  However, the IP address of your server is private and cannot be located from a remote location.  Therefore, the time on the phone will begin blinking 12:00 and/or set to some other incorrect time.<br />
<br />
You can manually configure the NTP settings on your Polycom or Aastra phone if you follow these directions:<br />
<br />
Aastra<br />
<br />
   1. Find the IP Address of your phone<br />
   2. Open your web browser and navigate to that IP address (type it into the URL field)<br />
   3. Login with username admin and password 22222<br />
   4. Click on Network Settings<br />
   5. Set Time Server 1 to pool.ntp.org<br />
   6. Click Save Settings<br />
   7. Click on Reset<br />
   8. Click on Restart Phone<br />
<br />
Polycom<br />
<br />
   1. Find the IP Address of your phone<br />
   2. Open your web browser and navigate to that IP address (type it into the URL field)<br />
   3. Login with username Polycom and password 456<br />
   4. Click on General<br />
   5. Set the SNTP Server to pool.ntp.org<br />
   6. Click Submit<br />
<br />
As long as the router in your remote location allows the phone to communicate with pool.ntp.org, the phone should correctly synchronize with the local time.]]></contents>
<contentstext><![CDATA[Phones local to the trixbox Pro use a Network Time Service in order to synchronize with the accurate, local time.  This Time Service comes in the form of an IP address given to the phone by the trixbox Pro (If used as a DHCP server) or by your own DHCP server.  

This technology is called NTP, and though Fonality tries to automate the process of a new phone booting up and correctly synchronizing the time for your location, when the phone is moved from the office to a remote location (including a remote office that does not have a trixbox Pro installed on-site), the automated process will fail.

Fonality uses the IP address of your trixbox Pro as the NTP value for every phone.  When the phone is moved to a remote location, it stores this IP address.  However, the IP address of your server is private and cannot be located from a remote location.  Therefore, the time on the phone will begin blinking 12:00 and/or set to some other incorrect time.

You can manually configure the NTP settings on your Polycom or Aastra phone if you follow these directions:

Aastra

   1. Find the IP Address of your phone
   2. Open your web browser and navigate to that IP address (type it into the URL field)
   3. Login with username admin and password 22222
   4. Click on Network Settings
   5. Set Time Server 1 to pool.ntp.org
   6. Click Save Settings
   7. Click on Reset
   8. Click on Restart Phone

Polycom

   1. Find the IP Address of your phone
   2. Open your web browser and navigate to that IP address (type it into the URL field)
   3. Login with username Polycom and password 456
   4. Click on General
   5. Set the SNTP Server to pool.ntp.org
   6. Click Submit

As long as the router in your remote location allows the phone to communicate with pool.ntp.org, the phone should correctly synchronize with the local time.]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[Setting the Time on Polycom and Asstra Phones with Trixbox Pro]]></subject>
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<kbarticle>
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<contents><![CDATA[These steps should work for all Aastra Wireless Handsets. <br />
<br />
Keep in mind, there is a limitation of 4 handsets per base station.<br />
<br />
During this process, keep the handset within 3 feet of the base station.<br />
<br />
On the base station:<br />
<br />
   1. Press Options<br />
   2. Select Handset Pairing.  (For a 57i CT, Select Preferences, then Handset Pairing)<br />
   3. Choose Pair and Press Select.<br />
<br />
On the Handset:<br />
<br />
   1. Press the Menu key<br />
   2. Select Hset pairing<br />
   3. Press the Pair soft key<br />
<br />
<br />
The base station and handset will attempt to connect, and then will display â€œPairing successfulâ€ or â€œPairing failedâ€.<br />
<br />
If the pairing fails, remove the handset battery. Wait 5 minutes, replace the handset battery and try again. It may take a few tries to pair successfully.  If the pairing continues to fail, you may have a bad unit.  In this event, you should contact your distributor/reseller for troubleshooting purposes.  <br />
]]></contents>
<contentstext><![CDATA[These steps should work for all Aastra Wireless Handsets. 

Keep in mind, there is a limitation of 4 handsets per base station.

During this process, keep the handset within 3 feet of the base station.

On the base station:

   1. Press Options
   2. Select Handset Pairing.  (For a 57i CT, Select Preferences, then Handset Pairing)
   3. Choose Pair and Press Select.

On the Handset:

   1. Press the Menu key
   2. Select Hset pairing
   3. Press the Pair soft key


The base station and handset will attempt to connect, and then will display â€œPairing successfulâ€ or â€œPairing failedâ€.

If the pairing fails, remove the handset battery. Wait 5 minutes, replace the handset battery and try again. It may take a few tries to pair successfully.  If the pairing continues to fail, you may have a bad unit.  In this event, you should contact your distributor/reseller for troubleshooting purposes.  
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Pair an Aastra Handset with a Base Station]]></subject>
<isedited><![CDATA[0]]></isedited>
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<contents><![CDATA[Reset Local Config to factory Default<br />
--------------------------------<br />
Step 1: Press the 'Menu' key.<br />
Step 2: Select 'Settings'.<br />
Step 3: Select 'Advanced'.<br />
Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.<br />
Step 5: Select 'Admin Settings'.<br />
Step 6: Select 'Reset to Default'.<br />
Step 7: Select 'Reset Local Config'.<br />
<br />
<br />
Reset Device Settings to Factory Default<br />
-----------------------------------<br />
IP 30x: Hold down the 4, 6, 8 and * keys simultaneously.<br />
IP 430: Hold down the 1, 3, 5 and 7 keys simultaneously.<br />
IP 50x: Hold down the 4, 6, 8 and * keys simultaneously.<br />
IP 60x: Hold down the 4, 6, 8 and * keys simultaneously.<br />
<br />
<br />
Format File System for a Fresh Firmware Load<br />
-----------------------------------<br />
Step 1: Press the 'Menu' hard key.<br />
Step 2: Select 'Settings'.<br />
Step 3: Select 'Advanced'.<br />
Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.<br />
Step 5: Select 'Admin Settings'.<br />
Step 6: Select 'Reset to Default'.<br />
Step 7: Select 'Format File System'. <br />
<br />
NOTE: This method will require the Administrator Password(456)<br />
]]></contents>
<contentstext><![CDATA[Reset Local Config to factory Default
--------------------------------
Step 1: Press the 'Menu' key.
Step 2: Select 'Settings'.
Step 3: Select 'Advanced'.
Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.
Step 5: Select 'Admin Settings'.
Step 6: Select 'Reset to Default'.
Step 7: Select 'Reset Local Config'.


Reset Device Settings to Factory Default
-----------------------------------
IP 30x: Hold down the 4, 6, 8 and * keys simultaneously.
IP 430: Hold down the 1, 3, 5 and 7 keys simultaneously.
IP 50x: Hold down the 4, 6, 8 and * keys simultaneously.
IP 60x: Hold down the 4, 6, 8 and * keys simultaneously.


Format File System for a Fresh Firmware Load
-----------------------------------
Step 1: Press the 'Menu' hard key.
Step 2: Select 'Settings'.
Step 3: Select 'Advanced'.
Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.
Step 5: Select 'Admin Settings'.
Step 6: Select 'Reset to Default'.
Step 7: Select 'Format File System'. 

NOTE: This method will require the Administrator Password(456)
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Restore Factory Defaults/Format a Polycom Phone]]></subject>
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<contents><![CDATA[The process for installing trixbox CE on an appliance is very similar to the normal install process.<br />
<br />
1.) Download the trixbox CE ISO (http://www.trixbox.org/downloads)<br />
<br />
2.) Burn the ISO image to a CD.  Insert the CD into an optical drive that is connected to your appliance.<br />
<br />
3.) Boot the appliance from the CD drive.  This will bring you to the trixbox CE installation screen.  <br />
<br />
4.) Normally at this point, you would use the "install" command.  But due to the Appliance needing specific drivers for the hardware and LCD screen, you will issue the "appliance" command.  <br />
<br />
5.) The install process will begin.  This is an entirely scripted process, and does not require any further steps.  It will take between five and ten minutes to install.<br />
<br />
The entire process is the same as the normal CE install, with the sole difference being the "appliance" command.  <br />
]]></contents>
<contentstext><![CDATA[The process for installing trixbox CE on an appliance is very similar to the normal install process.

1.) Download the trixbox CE ISO (http://www.trixbox.org/downloads)

2.) Burn the ISO image to a CD.  Insert the CD into an optical drive that is connected to your appliance.

3.) Boot the appliance from the CD drive.  This will bring you to the trixbox CE installation screen.  

4.) Normally at this point, you would use the "install" command.  But due to the Appliance needing specific drivers for the hardware and LCD screen, you will issue the "appliance" command.  

5.) The install process will begin.  This is an entirely scripted process, and does not require any further steps.  It will take between five and ten minutes to install.

The entire process is the same as the normal CE install, with the sole difference being the "appliance" command.  
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to Install trixbox CE on a trixbox Appliance]]></subject>
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<contents><![CDATA[The Aastra 6730i and 6731i model IP phones come with 8 programmable keys for customization, which are located at the top of the phone.  These keys can be set as BLFs (Busy Lamp Fields), Speed Dials, and XML application commands amongst other settings.  But as anyone who has used these phone know, out of those keys, two come from factory locked, with pre-configured settings.  The number 5 and 6 keys are set as â€œSaveâ€ and â€œDeleteâ€ by default, which can be used in conjunction with the XML phone directory.  But for those of us who either donâ€™t use XML, or wish to be able to configure the phone to their needs, we have to edit the phoneâ€™s config to unlock those keys for editing.<br />
<br />
To do this, you must edit your phoneâ€™s config file to contain the following:<br />
<br />
prgkey1 locked: 0<br />
prgkey2 locked: 0<br />
<br />
Setting these fields to â€˜0â€² unlocks the buttons from their pre-config, and  a â€˜1â€² will return them to a locked state.  So if you edit your config to â€˜0â€², and then load it into the phone using a TFTP Server, you will then gain access to those keys through your phoneâ€™s web GUI.]]></contents>
<contentstext><![CDATA[The Aastra 6730i and 6731i model IP phones come with 8 programmable keys for customization, which are located at the top of the phone.  These keys can be set as BLFs (Busy Lamp Fields), Speed Dials, and XML application commands amongst other settings.  But as anyone who has used these phone know, out of those keys, two come from factory locked, with pre-configured settings.  The number 5 and 6 keys are set as â€œSaveâ€ and â€œDeleteâ€ by default, which can be used in conjunction with the XML phone directory.  But for those of us who either donâ€™t use XML, or wish to be able to configure the phone to their needs, we have to edit the phoneâ€™s config to unlock those keys for editing.

To do this, you must edit your phoneâ€™s config file to contain the following:

prgkey1 locked: 0
prgkey2 locked: 0

Setting these fields to â€˜0â€² unlocks the buttons from their pre-config, and  a â€˜1â€² will return them to a locked state.  So if you edit your config to â€˜0â€², and then load it into the phone using a TFTP Server, you will then gain access to those keys through your phoneâ€™s web GUI.]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How to unlock Aastra 6730i/6731i programmable keys]]></subject>
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<contents><![CDATA[   1. 1. Provisional 1xx<br />
         1. 1.1. 100 Trying<br />
         2. 1.2. 180 Ringing<br />
         3. 1.3. 181 Call Is Being Forwarded<br />
         4. 1.4. 182 Queued<br />
         5. 1.5. 183 Session Progress<br />
   2. 2. Successful 2xx<br />
         1. 2.1. 200 OK<br />
   3. 3. Redirection 3xx<br />
         1. 3.1. 300 Multiple Choices<br />
         2. 3.2. 301 Moved Permanently<br />
         3. 3.3. 302 Moved Temporarily<br />
         4. 3.4. 305 Use Proxy<br />
         5. 3.5. 380 Alternative Service<br />
   4. 4. Request Failure 4xx<br />
         1. 4.1. 400 Bad Request<br />
         2. 4.2. 401 Unauthorized<br />
         3. 4.3. 402 Payment Required<br />
         4. 4.4. 403 Forbidden<br />
         5. 4.5. 404 Not Found<br />
         6. 4.6. 405 Method Not Allowed<br />
         7. 4.7. 406 Not Acceptable<br />
         8. 4.8. 407 Proxy Authentication Required<br />
         9. 4.9. 408 Request Timeout<br />
        10. 4.10. 410 Gone<br />
        11. 4.11. 413 Request Entity Too Large<br />
        12. 4.12. 414 Request-URI Too Long<br />
        13. 4.13. 415 Unsupported Media Type<br />
        14. 4.14. 416 Unsupported URI Scheme<br />
        15. 4.15. 420 Bad Extension<br />
        16. 4.16. 421 Extension Required<br />
        17. 4.17. 423 Interval Too Brief<br />
        18. 4.18. 480 Temporarily Unavailable<br />
        19. 4.19. 481 Call/Transaction Does Not Exist<br />
        20. 4.20. 482 Loop Detected<br />
        21. 4.21. 483 Too Many Hops<br />
        22. 4.22. 484 Address Incomplete<br />
        23. 4.23. 485 Ambiguous<br />
        24. 4.24. 486 Busy Here<br />
        25. 4.25. 487 Request Terminated<br />
        26. 4.26. 488 Not Acceptable Here<br />
        27. 4.27. 491 Request Pending<br />
        28. 4.28. 493 Undecipherable<br />
   5. 5. Server Failure 5xx<br />
         1. 5.1. 500 Server Internal Error<br />
         2. 5.2. 501 Not Implemented<br />
         3. 5.3. 502 Bad Gateway<br />
         4. 5.4. 503 Service Unavailable<br />
         5. 5.5. 504 Server Time-out<br />
         6. 5.6. 505 Version Not Supported<br />
         7. 5.7. 513 Message Too Large<br />
   6. 6. Global Failures 6xx<br />
         1. 6.1. 600 Busy Everywhere<br />
         2. 6.2. 603 Decline<br />
         3. 6.3. 604 Does Not Exist Anywhere<br />
         4. 6.4. 606 Not Acceptable]]></contents>
<contentstext><![CDATA[   1. 1. Provisional 1xx
         1. 1.1. 100 Trying
         2. 1.2. 180 Ringing
         3. 1.3. 181 Call Is Being Forwarded
         4. 1.4. 182 Queued
         5. 1.5. 183 Session Progress
   2. 2. Successful 2xx
         1. 2.1. 200 OK
   3. 3. Redirection 3xx
         1. 3.1. 300 Multiple Choices
         2. 3.2. 301 Moved Permanently
         3. 3.3. 302 Moved Temporarily
         4. 3.4. 305 Use Proxy
         5. 3.5. 380 Alternative Service
   4. 4. Request Failure 4xx
         1. 4.1. 400 Bad Request
         2. 4.2. 401 Unauthorized
         3. 4.3. 402 Payment Required
         4. 4.4. 403 Forbidden
         5. 4.5. 404 Not Found
         6. 4.6. 405 Method Not Allowed
         7. 4.7. 406 Not Acceptable
         8. 4.8. 407 Proxy Authentication Required
         9. 4.9. 408 Request Timeout
        10. 4.10. 410 Gone
        11. 4.11. 413 Request Entity Too Large
        12. 4.12. 414 Request-URI Too Long
        13. 4.13. 415 Unsupported Media Type
        14. 4.14. 416 Unsupported URI Scheme
        15. 4.15. 420 Bad Extension
        16. 4.16. 421 Extension Required
        17. 4.17. 423 Interval Too Brief
        18. 4.18. 480 Temporarily Unavailable
        19. 4.19. 481 Call/Transaction Does Not Exist
        20. 4.20. 482 Loop Detected
        21. 4.21. 483 Too Many Hops
        22. 4.22. 484 Address Incomplete
        23. 4.23. 485 Ambiguous
        24. 4.24. 486 Busy Here
        25. 4.25. 487 Request Terminated
        26. 4.26. 488 Not Acceptable Here
        27. 4.27. 491 Request Pending
        28. 4.28. 493 Undecipherable
   5. 5. Server Failure 5xx
         1. 5.1. 500 Server Internal Error
         2. 5.2. 501 Not Implemented
         3. 5.3. 502 Bad Gateway
         4. 5.4. 503 Service Unavailable
         5. 5.5. 504 Server Time-out
         6. 5.6. 505 Version Not Supported
         7. 5.7. 513 Message Too Large
   6. 6. Global Failures 6xx
         1. 6.1. 600 Busy Everywhere
         2. 6.2. 603 Decline
         3. 6.3. 604 Does Not Exist Anywhere
         4. 6.4. 606 Not Acceptable]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[Sip Response Codes]]></subject>
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<contents><![CDATA[If you can match the impedance setting expected by your country's PSTN, you will have much better results in detecting the proper state of your PSTN line as well as have better audio quality.<br />
<br />
The following settings apply to various countries:<br />
<br />
    * 600 ohms (US/Canada)<br />
    * 900 ohms (US/Canada)<br />
    * 220+820//115nF (Germany)<br />
    * 200+680//100nF (China)<br />
    * 370+620//310nF (UK)<br />
    * 180+910//150nF (France)<br />
    * 270+750//150nF (Sweden)<br />
    * 120+820//110nF (Norway)<br />
    * 150+830//72nF (Belgium)<br />
    * 400+500//330nF (Denmark)<br />
]]></contents>
<contentstext><![CDATA[If you can match the impedance setting expected by your country's PSTN, you will have much better results in detecting the proper state of your PSTN line as well as have better audio quality.

The following settings apply to various countries:

    * 600 ohms (US/Canada)
    * 900 ohms (US/Canada)
    * 220+820//115nF (Germany)
    * 200+680//100nF (China)
    * 370+620//310nF (UK)
    * 180+910//150nF (France)
    * 270+750//150nF (Sweden)
    * 120+820//110nF (Norway)
    * 150+830//72nF (Belgium)
    * 400+500//330nF (Denmark)
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[What impedence setting should I choose for my SPA3102 FXO port?]]></subject>
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<contents><![CDATA[There are 3 methods to manually add a device.  The instructions for Support Local Phone apply to Unsupport Phones as well. <br />
<br />
SIP DEVICE PASSWORDS ARE AUTOMATICALLY GENERATED.<br />
Supported Local Phone<br />
<br />
   1. Within your Admin Panel, navigate to Extensions --> phones<br />
   2. Fill in the MAC and select the Vendor<br />
   3. If the phone is a Supported phone, when you select the Vendor, the Desc field will auto-populate<br />
   4. If the phone does not match any Vendor, choose Other and add your own Desc<br />
   5. Click "Add Device" below<br />
   6. Now, set the tftp server on the phone to: "s[server_id].trixbox.fonality.com"<br />
   7. Reboot your phone<br />
   8. Now click on Extensions in your Admin Panel and select the proper extension for this phone (if you haven't created the extension yet, please read How do I add an extension?)<br />
   9. Select your newly created device from the list - devices that are already paired with an extension will display the extension in ( ) as pictured:<br />
<br />
Supported Remote Phone<br />
<br />
   1. Fill in the MAC and Vendor below.<br />
   2. Click "Add Device" below.<br />
   3. Now, set the tftp server on the phone to: "s[server_id]x.trixbox.fonality.com". Did you notice the x? :)<br />
   4. Reboot the phone.<br />
   5. If your phone cannot get its configuration from your [server_brand] server then:<br />
         1. a. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"<br />
         2. Set the SIP username and password<br />
         3. Reboot your phone.<br />
<br />
Unsupported Phone (local or remote)<br />
<br />
   1. Fill in the MAC and Vendor below.<br />
   2. Click "Add Device" below.<br />
   3. If LOCAL:<br />
         1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id].trixbox.fonality.com"<br />
   4. If REMOTE:<br />
         1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"<br />
         2. Note the x!<br />
         3. Set the SIP username and password<br />
   5. Reboot your phone.<br />
]]></contents>
<contentstext><![CDATA[There are 3 methods to manually add a device.  The instructions for Support Local Phone apply to Unsupport Phones as well. 

SIP DEVICE PASSWORDS ARE AUTOMATICALLY GENERATED.
Supported Local Phone

   1. Within your Admin Panel, navigate to Extensions --> phones
   2. Fill in the MAC and select the Vendor
   3. If the phone is a Supported phone, when you select the Vendor, the Desc field will auto-populate
   4. If the phone does not match any Vendor, choose Other and add your own Desc
   5. Click "Add Device" below
   6. Now, set the tftp server on the phone to: "s[server_id].trixbox.fonality.com"
   7. Reboot your phone
   8. Now click on Extensions in your Admin Panel and select the proper extension for this phone (if you haven't created the extension yet, please read How do I add an extension?)
   9. Select your newly created device from the list - devices that are already paired with an extension will display the extension in ( ) as pictured:

Supported Remote Phone

   1. Fill in the MAC and Vendor below.
   2. Click "Add Device" below.
   3. Now, set the tftp server on the phone to: "s[server_id]x.trixbox.fonality.com". Did you notice the x? :)
   4. Reboot the phone.
   5. If your phone cannot get its configuration from your [server_brand] server then:
         1. a. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"
         2. Set the SIP username and password
         3. Reboot your phone.

Unsupported Phone (local or remote)

   1. Fill in the MAC and Vendor below.
   2. Click "Add Device" below.
   3. If LOCAL:
         1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id].trixbox.fonality.com"
   4. If REMOTE:
         1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"
         2. Note the x!
         3. Set the SIP username and password
   5. Reboot your phone.
]]></contentstext>
<categories>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[8]]></creatorid>
<author><![CDATA[Matthew Hanson]]></author>
<email><![CDATA[matthew@888voipstore.com]]></email>
<subject><![CDATA[How do I add a phone/device to trixbox PRO?]]></subject>
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<contents><![CDATA[Data Sheet for Linksys Analog Adapter PAP2T<br /><br />Link: <a href="http://www.888voipstore.com/downloads/linksys/PAP2admin.pdf" target="_blank">http://www.888voipstore.com/downloads/linksys/PAP2admin.pdf</a>]]></contents>
<contentstext><![CDATA[Data Sheet for Linksys Analog Adapter PAP2T]]></contentstext>
<categories>
<categoryid><![CDATA[3]]></categoryid>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[0]]></creatorid>
<author><![CDATA[- NA -]]></author>
<email />
<subject><![CDATA[PAP2T Data Sheet]]></subject>
<isedited><![CDATA[0]]></isedited>
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<kbarticle>
<kbarticleid><![CDATA[35]]></kbarticleid>
<contents><![CDATA[Userguide for Sipura / PAP2 products]]></contents>
<contentstext><![CDATA[Userguide for Sipura / PAP2 products]]></contentstext>
<categories>
<categoryid><![CDATA[3]]></categoryid>
</categories>
<creator><![CDATA[2]]></creator>
<creatorid><![CDATA[0]]></creatorid>
<author><![CDATA[- NA -]]></author>
<email />
<subject><![CDATA[PAP2T Userguide]]></subject>
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<kbarticle>
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<contents><![CDATA[SoundPoint IP / SoundStation IP Administrator's Guide, Version 2.0.x <br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/polycom-sip2.0.x-admin-guide.pdf" target="_blank">http://www.888voipstore.com/downloads/polycom/polycom-sip2.0.x-admin-guide.pdf</a>]]></contents>
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<contents><![CDATA[SoundPoint IP / SoundStation IP BootROM 3.2.3<br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/BootRom_3_2_3_release_sig.zip" target="_blank">http://www.888voipstore.com/downloads/polycom/BootRom_3_2_3_release_sig.zip</a>]]></contents>
<contentstext><![CDATA[SoundPoint IP / SoundStation IP BootROM 3.2.3]]></contentstext>
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<contents><![CDATA[Polycom IP601/650 Expansion Module Quick Start Guide<br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/soundpoint_ip_expansion_module_quick_start_guide.pdf" target="_blank">http://www.888voipstore.com/downloads/polycom/soundpoint_ip_expansion_module_quick_start_guide.pdf</a>]]></contents>
<contentstext><![CDATA[Polycom IP601/650 Expansion Module Quick Start Guide]]></contentstext>
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<contents><![CDATA[Mediant1000UserDocs.zip<br />
<br /><br />Link: <a href="http://www.888voipstore.com/downloads/audiocodes/Mediant1000UserDocs.zip" target="_blank">http://www.888voipstore.com/downloads/audiocodes/Mediant1000UserDocs.zip</a>]]></contents>
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<contents><![CDATA[M1000_V5_BASE.zip<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/M1000_V5_BASE.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/M1000_V5_BASE.zip</a>]]></contents>
<contentstext><![CDATA[M1000_V5_BASE.zip

Link: http://www.888voip.com/downloads/audiocodes/M1000_V5_BASE.zip
[1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/M1000_V5_BASE.zip
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<contents><![CDATA[M1000_SIP_F5.00A.036.003.cmp firmware for Audiocodes Mediant 1000<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.00A.036.003.cmp" target="_blank">http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.00A.036.003.cmp</a>]]></contents>
<contentstext><![CDATA[M1000_SIP_F5.00A.036.003.cmp firmware for Audiocodes Mediant 1000

Link:
http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.00A.036.003.cmp
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.00A.036.003.cmp
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<contents><![CDATA[MP118_SIP_F5.00A.042.001.cmp firmware for Audiocodes Mediapack MPxxx series voip gateways<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.00A.042.001.cmp" target="_blank">http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.00A.042.001.cmp</a>]]></contents>
<contentstext><![CDATA[MP118_SIP_F5.00A.042.001.cmp firmware for Audiocodes Mediapack MPxxx
series voip gateways

Link:
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.00A.042.001.cmp
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.00A.042.001.cmp
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<contents><![CDATA[LTRT-65408 MP-11x & MP-124 SIP User's Manual Ver 5.0.pdf<br />
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<contentstext><![CDATA[LTRT-65408 MP-11x & MP-124 SIP User's Manual Ver 5.0.pdf

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<contents><![CDATA[LTRT-66507 CPE Configuration Guide for IP Voice Mail Ver 5.2.pdf<br /><br />Link: <a href="http://www.audiocodes.com/asp/DoDownload2.asp?ID=5781" target="_blank">http://www.audiocodes.com/asp/DoDownload2.asp?ID=5781</a>]]></contents>
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<contents><![CDATA[Audiocodes 5.2 SIP firmware for the Mediant1000<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/M1000_V52_BASE.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/M1000_V52_BASE.zip</a>]]></contents>
<contentstext><![CDATA[Audiocodes 5.2 SIP firmware for the Mediant1000

Link: http://www.888voip.com/downloads/audiocodes/M1000_V52_BASE.zip
[1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/M1000_V52_BASE.zip
]]></contentstext>
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<contents><![CDATA[Linksys SPA products Admin Guide<br /><br />Link: <a href="http://www.888voipstore.com/downloads/linksys/SpaAdminGuide.pdf" target="_blank">http://www.888voipstore.com/downloads/linksys/SpaAdminGuide.pdf</a>]]></contents>
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<contents><![CDATA[Audiocodes MP118 - MP124 5.2 Firmware<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/SIP_MP-11x_and_MP-124_version 5.2.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/SIP_MP-11x_and_MP-124_version 5.2.zip</a>]]></contents>
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Link:
http://www.888voip.com/downloads/audiocodes/SIP_MP-11x_and_MP-124_version
5.2.zip [1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/SIP_MP-11x_and_MP-124_version
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[1]

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------
[1]
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<contents><![CDATA[Release Notes SIP Application SoundPoint and SoundStation IP Version 3.0.3 RevB 13 June 2008]]></contents>
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<contents><![CDATA[Polycom Firware SoundPoint IP and SoundStation IP SIP 3.0.3 Rev B  (81.38MB ZIP)<br />
<br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/spip_ssip_3_0_3RevB_release_sig.zip" target="_blank">http://www.888voipstore.com/downloads/polycom/spip_ssip_3_0_3RevB_release_sig.zip</a>]]></contents>
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<contents><![CDATA[HT286 (Rev. 3.0)<br />
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v1.1.0.31
v1.1.0.26]]></contentstext>
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<contents><![CDATA[v1.0.1.41<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXE5024_GXE5028/firmware/Release_GXE5000_1.0.1.41.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXE5024_GXE5028/firmware/Release_GXE5000_1.0.1.41.zip</a>]]></contents>
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<contents><![CDATA[v1.0.1.35<br />
<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_HT502_1.0.1.35.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_HT502_1.0.1.35.zip</a>]]></contents>
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<contents><![CDATA[v1.0.1.35<br />
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<contents><![CDATA[v1.1.6.46<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/BT200_GXP/Release_BT200_GXP_1.1.6.46.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/BT200_GXP/Release_BT200_GXP_1.1.6.46.zip</a>]]></contents>
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<contents><![CDATA[v1.1.3.50<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXV-3000/Release_GXV3000_1.1.3.50.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXV-3000/Release_GXV3000_1.1.3.50.zip</a>]]></contents>
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<contents><![CDATA[v1.0.1.35<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_GXW4004_1.0.1.35.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_GXW4004_1.0.1.35.zip</a>]]></contents>
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<contents><![CDATA[v1.0.1.35<br /><br />Link: <a href="http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_GXW4008_1.0.1.35.zip" target="_blank">http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXW40xx_HT50x/Release_GXW4008_1.0.1.35.zip</a>]]></contents>
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<contents><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2875 KB <br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/6757i_FC-001088-06-REV00_sr_2.4.0.96_0811.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/6757i_FC-001088-06-REV00_sr_2.4.0.96_0811.zip</a>]]></contents>
<contentstext><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2875 KB ]]></contentstext>
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<subject><![CDATA[Aastra 57i SIP, Version 2.4.0]]></subject>
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<contents><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2921 KB<br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/6757iCT_FC-001089-06-REV00_sr_2.4.0.96_0811.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/6757iCT_FC-001089-06-REV00_sr_2.4.0.96_0811.zip</a>]]></contents>
<contentstext><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2921 KB]]></contentstext>
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<subject><![CDATA[Aastra  	  57i CT SIP, Version 2.4.0]]></subject>
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<contents><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2780 KB <br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9143i_FC-001092-04-REV00_sr_2.4.0.96_0811.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9143i_FC-001092-04-REV00_sr_2.4.0.96_0811.zip</a>]]></contents>
<contentstext><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2780 KB ]]></contentstext>
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<subject><![CDATA[Aastra 9143i SIP, Version 2.4.0]]></subject>
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<contents><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2803 KB <br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9480i_FC-001097-04-REV00_sr_2.4.0.96_0811.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9480i_FC-001097-04-REV00_sr_2.4.0.96_0811.zip</a>]]></contents>
<contentstext><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2803 KB ]]></contentstext>
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<subject><![CDATA[Aastra 9480i SIP, Version 2.4.0]]></subject>
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<contents><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2845 KB <br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9480iCT_FC-001101-04-REV00_sr_2.4.0.96_0811.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-F8A9A6D4/03/9480iCT_FC-001101-04-REV00_sr_2.4.0.96_0811.zip</a>]]></contents>
<contentstext><![CDATA[Version 2.4.0, November 2008 , English , ZIP , 2845 KB ]]></contentstext>
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<contents><![CDATA[Polycom Firmware 3.1 RevC <br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/spip_ssip_3_1_0RevC_release_sig.zip" target="_blank">http://www.888voipstore.com/downloads/polycom/spip_ssip_3_1_0RevC_release_sig.zip</a>]]></contents>
<contentstext><![CDATA[Polycom Firmware 3.1 RevC ]]></contentstext>
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<contents><![CDATA[SoundPoint IP and SoundStation IP SIP Firmware 3.1.1<br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/spip_ssip_3_1_1_release_sig.zip" target="_blank">http://www.888voipstore.com/downloads/polycom/spip_ssip_3_1_1_release_sig.zip</a>]]></contents>
<contentstext><![CDATA[SoundPoint IP and SoundStation IP SIP Firmware 3.1.1]]></contentstext>
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<subject><![CDATA[Polycom SoundPoint IP and SoundStation IP SIP 3.1.1]]></subject>
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<contents><![CDATA[December 2008<br /><br />Link: <a href="http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-534FACC8/03/SIP-DECT_OMM_V1.6.6_0812.zip" target="_blank">http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-534FACC8/03/SIP-DECT_OMM_V1.6.6_0812.zip</a>]]></contents>
<contentstext><![CDATA[December 2008]]></contentstext>
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<creator><![CDATA[2]]></creator>
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<subject><![CDATA[SIP DECT Software V. 1.6.6]]></subject>
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<contents><![CDATA[Version 4.61<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/SIP-MP-1xx-version-4.61.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/SIP-MP-1xx-version-4.61.zip</a>]]></contents>
<contentstext><![CDATA[Version 4.61

Link:
http://www.888voip.com/downloads/audiocodes/SIP-MP-1xx-version-4.61.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/SIP-MP-1xx-version-4.61.zip
]]></contentstext>
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<contents><![CDATA[SIP 5.4 Build A<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.40.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.40.zip</a>]]></contents>
<contentstext><![CDATA[SIP 5.4 Build A

Link: http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.40.zip
[1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/M1000_SIP_F5.40.zip
]]></contentstext>
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<subject><![CDATA[Audiocodes Mediant 1000 Firmware Update]]></subject>
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<contents><![CDATA[Firmware update to allow enable H.323 on Audiocodes Devices.<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/MP11xandMP124H.323version5.0.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/MP11xandMP124H.323version5.0.zip</a>]]></contents>
<contentstext><![CDATA[Firmware update to allow enable H.323 on Audiocodes Devices.

Link:
http://www.888voip.com/downloads/audiocodes/MP11xandMP124H.323version5.0.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/MP11xandMP124H.323version5.0.zip
]]></contentstext>
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<contents><![CDATA[<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.024.003.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.024.003.zip</a>]]></contents>
<contentstext><![CDATA[

Link:
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.024.003.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.024.003.zip
]]></contentstext>
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<contents><![CDATA[<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/BootP_2_3_0_5.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/BootP_2_3_0_5.zip</a>]]></contents>
<contentstext><![CDATA[

Link: http://www.888voip.com/downloads/audiocodes/BootP_2_3_0_5.zip
[1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/BootP_2_3_0_5.zip
]]></contentstext>
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<contents><![CDATA[<br /><br />Link: <a href="http://www.888voipstore.com/downloads/aastra/ConfigFiles/Aastra 6730i6731i Templates.zip" target="_blank">http://www.888voipstore.com/downloads/aastra/ConfigFiles/Aastra 6730i6731i Templates.zip</a>]]></contents>
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<contents><![CDATA[Audiocodes MP20X Firmware Update 2.6.4<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/MP20x_2_6_4.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/MP20x_2_6_4.zip</a>]]></contents>
<contentstext><![CDATA[Audiocodes MP20X Firmware Update 2.6.4

Link: http://www.888voip.com/downloads/audiocodes/MP20x_2_6_4.zip [1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/MP20x_2_6_4.zip
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<contents><![CDATA[v1.0.1<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/320HD v1.0.1.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/320HD v1.0.1.zip</a>]]></contents>
<contentstext><![CDATA[v1.0.1

Link: http://www.888voip.com/downloads/audiocodes/320HD v1.0.1.zip [1]

Links:
------
[1] http://www.888voip.com/downloads/audiocodes/320HD v1.0.1.zip
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<contents><![CDATA[Complete Polycom SIP 3.2.0 firmware for all current models of Polycom phones.<br /><br />Link: <a href="http://www.888voipstore.com/downloads/polycom/spip_ssip_vvx_3_2_0_release_sig_combined.zip" target="_blank">http://www.888voipstore.com/downloads/polycom/spip_ssip_vvx_3_2_0_release_sig_combined.zip</a>]]></contents>
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<contents><![CDATA[MP118_SIP_F5.60A.027.002 Firmware<br /><br />Link: <a href="http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.027.002.zip" target="_blank">http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.027.002.zip</a>]]></contents>
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Link:
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.027.002.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/audiocodes/MP118_SIP_F5.60A.027.002.zip
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Link:
http://www.888voip.com/downloads/polycom/New/spip_ssip_vvx_3_2_0.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/polycom/New/spip_ssip_vvx_3_2_0.zip
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Link: http://www.888voip.com/downloads/polycom/New/BootROM_4_1_2.zip
[1]

Links:
------
[1] http://www.888voip.com/downloads/polycom/New/BootROM_4_1_2.zip
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<contents><![CDATA[<br /><br />Link: <a href="http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip" target="_blank">http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip</a>]]></contents>
<contentstext><![CDATA[

Link:
http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
[1]

Links:
------
[1]
http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
]]></contentstext>
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<subject><![CDATA[Polycom UC 3.3 Split]]></subject>
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<contents><![CDATA[SIP-T28(P) V50 Firmware , Version 2.50.0.50, Released June 10, 2010<br /><br />Link: <a href="http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T28(P)%20V50%20Firmware-2.50.0.50.zip" target="_blank">http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T28(P)%20V50%20Firmware-2.50.0.50.zip</a>]]></contents>
<contentstext><![CDATA[SIP-T28(P) V50 Firmware , Version 2.50.0.50, Released June 10, 2010]]></contentstext>
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<contents><![CDATA[SIP-T26(P) V50 Firmware , Version 6.50.0.50, Released June 10, 2010<br /><br />Link: <a href="http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T26(P)%20V50%20Firmware-6.50.0.50.zip" target="_blank">http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T26(P)%20V50%20Firmware-6.50.0.50.zip</a>]]></contents>
<contentstext><![CDATA[SIP-T26(P) V50 Firmware , Version 6.50.0.50, Released June 10, 2010]]></contentstext>
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<contents><![CDATA[SIP-T22(P) V50 Firmware , Version 7.50.0.50, Released June 10, 2010<br /><br />Link: <a href="http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T22(P)%20V50%20Firmware-7.50.0.50.zip" target="_blank">http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T22(P)%20V50%20Firmware-7.50.0.50.zip</a>]]></contents>
<contentstext><![CDATA[SIP-T22(P) V50 Firmware , Version 7.50.0.50, Released June 10, 2010]]></contentstext>
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<contents><![CDATA[SIP-T20(P) V50 Firmware , Version 9.50.0.50, Released June 10, 2010<br /><br />Link: <a href="http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T20(P)%20V50%20Firmware-9.50.0.50.zip" target="_blank">http://www.888voipstore.com/downloads/yealink/Yealink%20SIP-T20(P)%20V50%20Firmware-9.50.0.50.zip</a>]]></contents>
<contentstext><![CDATA[SIP-T20(P) V50 Firmware , Version 9.50.0.50, Released June 10, 2010]]></contentstext>
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<contents><![CDATA[<p><strong>Getting Started</strong></p>
<p>In order to setup shared phonebooks, you will need to be able to create an XML file using a text editor such as Windows Notepad and be able to transfer this file to a web (HTTP) server.</p>
<p><strong>Basic&nbsp;Configuration</strong></p>
<p>Prepare the&nbsp;XML&nbsp;File.</p>
<p>The template below should be followed to create a file called phonebook.xml</p>
<p>&lt;?xml version="1.0" encoding="UTF-8" ?&gt;</p>
<p>&lt;xxxIPPhoneDirectory clearlight="true"&gt;</p>
<p>&lt;Title&gt;Phonelist&lt;/Title&gt;</p>
<p>&lt;Prompt&gt;Prompt&lt;/Prompt&gt;</p>
<p>&lt;DirectoryEntry&gt;</p>
<p>&lt;Name&gt;Support&lt;/Name&gt;</p>
<p>&lt;Telephone&gt;8881234567&lt;/Telephone&gt;</p>
<p>&lt;/DirectoryEntry&gt;</p>
<p>&lt;DirectoryEntry&gt;</p>
<p>&lt;Name&gt;Sales&lt;/Name&gt;</p>
<p>&lt;Telephone&gt;8887654321&lt;/Telephone&gt;</p>
<p>&lt;/DirectoryEntry&gt;</p>
<p>&lt;DirectoryEntry&gt;</p>
<p>&lt;Name&gt;Billing&lt;/Name&gt;</p>
<p>&lt;Telephone&gt;8881231231&lt;/Telephone&gt;</p>
<p>&lt;/DirectoryEntry&gt;</p>
<p>&lt;SoftKeyItem&gt;</p>
<p>&lt;Name&gt;#&lt;/Name&gt;</p>
<p>&lt;URL&gt;<a href="http://xxx/minibrowser/start.xml%3c/URL">http://xxx/minibrowser/start.xml&lt;/URL</a>&gt;</p>
<p>&lt;/SoftKeyItem&gt;</p>
<p>&lt;SoftKeyItem&gt;</p>
<p>&lt;Name&gt;*&lt;/Name&gt;</p>
<p>&lt;URL&gt;<a href="http://xxx/minibrowser/menu.xml%3c/URL">http://xxx/minibrowser/menu.xml&lt;/URL</a>&gt;</p>
<p>&lt;/SoftKeyItem&gt;</p>
<p>&lt;/xxxIPPhoneDirectory&gt;</p>
<p>The xxx value referred to above, is usually your company name, or the name of the server on your network hosting the finished xml file.</p>
<p>The &lt;Telephone&gt; number &lt;/Telephone&gt; field supports three formats,</p>
<ul>
<li>&lt;Telephone&gt;8881234567@sip.888voipstore.com&lt;/Telephone&gt;</li>
<li>&lt;Telephone&gt;8881234567&lt;/Telephone&gt;</li>
<li>&lt;Telephone&gt;sip:8881234567@sip.888voipstore.com&lt;/Telephone&gt;</li>
</ul>
<p>Each Contact, has a fixed format</p>
<ul>
<li>&lt;DirectoryEntry&gt;</li>
<li>&lt;Name&gt;Support&lt;/Name&gt;</li>
<li>&lt;Telephone&gt;8881234567@sip.888voipstore.com&lt;/Telephone&gt;</li>
<li>&lt;/DirectoryEntry&gt;</li>
</ul>
<p>Keep the format the same and therefore do not change anything except modifying the content between &lt;Name&gt; and &lt;/Name&gt; and also between &lt;Telephone&gt; and &lt;/Telephone&gt;. The complete section between &lt;Directory&gt; and &lt;/Directory&gt; defines just one contact.</p>
<p>For more contacts, add new sections in exactly the same format. When the file is complete, save it as phonebook.xml</p>
<p>Copy the XML&nbsp;file to your HTTP&nbsp;Server, using an FTP client or SFTP - upload the file into a world readable area on your webserver.</p>
<p><strong>Configuring your handset to download the phonebook using the management GUI</strong></p>
<p>Once logged into your phones web based interface, select Contacts followed by Remote phone book from the menu.</p>
<p>Enter the URL&nbsp;of where the phonebook.xml file is located , for example http://192.168.1.100/HTTP/phonebook.xml into the box under "Phone&nbsp;Book&nbsp;URL" and then enter a name under "Phone&nbsp;Book&nbsp;name", for example "Corporate&nbsp;Directory" and press&nbsp;Confirm.</p>
<p>On your handset, select Address Book and you will now see a new item called "Corporate", you can access your phone book by selecting "Corporate".</p>]]></contents>
<contentstext><![CDATA[Getting Started
In order to setup shared phonebooks, you will need to be able to create an XML file using a text editor such as Windows Notepad and be able to transfer this file to a web (HTTP) server.
Basic&nbsp;Configuration
Prepare the&nbsp;XML&nbsp;File.
The template below should be followed to create a file called phonebook.xml
&lt;?xml version="1.0" encoding="UTF-8" ?&gt;
&lt;xxxIPPhoneDirectory clearlight="true"&gt;
&lt;Title&gt;Phonelist&lt;/Title&gt;
&lt;Prompt&gt;Prompt&lt;/Prompt&gt;
&lt;DirectoryEntry&gt;
&lt;Name&gt;Support&lt;/Name&gt;
&lt;Telephone&gt;8881234567&lt;/Telephone&gt;
&lt;/DirectoryEntry&gt;
&lt;DirectoryEntry&gt;
&lt;Name&gt;Sales&lt;/Name&gt;
&lt;Telephone&gt;8887654321&lt;/Telephone&gt;
&lt;/DirectoryEntry&gt;
&lt;DirectoryEntry&gt;
&lt;Name&gt;Billing&lt;/Name&gt;
&lt;Telephone&gt;8881231231&lt;/Telephone&gt;
&lt;/DirectoryEntry&gt;
&lt;SoftKeyItem&gt;
&lt;Name&gt;#&lt;/Name&gt;
&lt;URL&gt;http://xxx/minibrowser/start.xml&lt;/URL&gt;
&lt;/SoftKeyItem&gt;
&lt;SoftKeyItem&gt;
&lt;Name&gt;*&lt;/Name&gt;
&lt;URL&gt;http://xxx/minibrowser/menu.xml&lt;/URL&gt;
&lt;/SoftKeyItem&gt;
&lt;/xxxIPPhoneDirectory&gt;
The xxx value referred to above, is usually your company name, or the name of the server on your network hosting the finished xml file.
The &lt;Telephone&gt; number &lt;/Telephone&gt; field supports three formats,

&lt;Telephone&gt;8881234567@sip.888voipstore.com&lt;/Telephone&gt;
&lt;Telephone&gt;8881234567&lt;/Telephone&gt;
&lt;Telephone&gt;sip:8881234567@sip.888voipstore.com&lt;/Telephone&gt;

Each Contact, has a fixed format

&lt;DirectoryEntry&gt;
&lt;Name&gt;Support&lt;/Name&gt;
&lt;Telephone&gt;8881234567@sip.888voipstore.com&lt;/Telephone&gt;
&lt;/DirectoryEntry&gt;

Keep the format the same and therefore do not change anything except modifying the content between &lt;Name&gt; and &lt;/Name&gt; and also between &lt;Telephone&gt; and &lt;/Telephone&gt;. The complete section between &lt;Directory&gt; and &lt;/Directory&gt; defines just one contact.
For more contacts, add new sections in exactly the same format. When the file is complete, save it as phonebook.xml
Copy the XML&nbsp;file to your HTTP&nbsp;Server, using an FTP client or SFTP - upload the file into a world readable area on your webserver.
Configuring your handset to download the phonebook using the management GUI
Once logged into your phones web based interface, select Contacts followed by Remote phone book from the menu.
Enter the URL&nbsp;of where the phonebook.xml file is located , for example http://192.168.1.100/HTTP/phonebook.xml into the box under "Phone&nbsp;Book&nbsp;URL" and then enter a name under "Phone&nbsp;Book&nbsp;name", for example "Corporate&nbsp;Directory" and press&nbsp;Confirm.
On your handset, select Address Book and you will now see a new item called "Corporate", you can access your phone book by selecting "Corporate".]]></contentstext>
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<subject><![CDATA[How to create an XML phonebook for Yealink Phones]]></subject>
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<contents><![CDATA[Using an SSH client, SSH or Telnet into the Patton's IP address<br /><br />Login into the unit. (Default: Username: administrator / Password: blank, no password)<br /><br />Once in the CLI (Command Line Interface), enter the following commands:<br /><br />1) enable <br />2) configure<br />3) debug context sip-gateway transport detail 5<br />4) debug context sip-gateway error detail 5<br />5) debug call-control detail 5<br />6) debug ccfxo<br />7) debug fxo<br /><br />Be sure to press enter after each command.<br /><br />This will allow you to log and capture all traffic through the Patton Device for debugging/troubleshooting purposes.&nbsp; <br />]]></contents>
<contentstext><![CDATA[Using an SSH client, SSH or Telnet into the Patton's IP address  Login into the unit. (Default: Username: administrator / Password: blank, no password)  Once in the CLI (Command Line Interface), enter the following commands:  1) enable  2) configure 3) debug context sip-gateway transport detail 5 4) debug context sip-gateway error detail 5 5) debug call-control detail 5 6) debug ccfxo 7) debug fxo  Be sure to press enter after each command.  This will allow you to log and capture all traffic through the Patton Device for debugging/troubleshooting purposes.&nbsp;  ]]></contentstext>
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<subject><![CDATA[How do I run a system debug on a Patton Gateway?]]></subject>
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<contents><![CDATA[There are 3 methods to manually add a device.&nbsp; The instructions for Support Local Phone apply to Unsupport Phones as well. <br /><br />SIP DEVICE PASSWORDS ARE AUTOMATICALLY GENERATED.<br />Supported Local Phone<br /><br />&nbsp;&nbsp; 1. Within your Admin Panel, navigate to Extensions --&gt; phones<br />&nbsp;&nbsp; 2. Fill in the MAC and select the Vendor<br />&nbsp;&nbsp; 3. If the phone is a Supported phone, when you select the Vendor, the Desc field will auto-populate<br />&nbsp;&nbsp; 4. If the phone does not match any Vendor, choose Other and add your own Desc<br />&nbsp;&nbsp; 5. Click "Add Device" below<br />&nbsp;&nbsp; 6. Now, set the tftp server on the phone to: "s[server_id].trixbox.fonality.com"<br />&nbsp;&nbsp; 7. Reboot your phone<br />&nbsp;&nbsp; 8. Now click on Extensions in your Admin Panel and select the proper extension for this phone (if you haven't created the extension yet, please read How do I add an extension?)<br />&nbsp;&nbsp; 9. Select your newly created device from the list - devices that are already paired with an extension will display the extension in ( ) as pictured:<br /><br />Supported Remote Phone<br /><br />&nbsp;&nbsp; 1. Fill in the MAC and Vendor below.<br />&nbsp;&nbsp; 2. Click "Add Device" below.<br />&nbsp;&nbsp; 3. Now, set the tftp server on the phone to: "s[server_id]x.trixbox.fonality.com". Did you notice the x? :)<br />&nbsp;&nbsp; 4. Reboot the phone.<br />&nbsp;&nbsp; 5. If your phone cannot get its configuration from your [server_brand] server then:<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. a. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Set the SIP username and password<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Reboot your phone.<br /><br />Unsupported Phone (local or remote)<br /><br />&nbsp;&nbsp; 1. Fill in the MAC and Vendor below.<br />&nbsp;&nbsp; 2. Click "Add Device" below.<br />&nbsp;&nbsp; 3. If LOCAL:<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id].trixbox.fonality.com"<br />&nbsp;&nbsp; 4. If REMOTE:<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com"<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Note the x!<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Set the SIP username and password<br />&nbsp;&nbsp; 5. Reboot your phone.<br />]]></contents>
<contentstext><![CDATA[There are 3 methods to manually add a device.&nbsp; The instructions for Support Local Phone apply to Unsupport Phones as well.   SIP DEVICE PASSWORDS ARE AUTOMATICALLY GENERATED. Supported Local Phone  &nbsp;&nbsp; 1. Within your Admin Panel, navigate to Extensions --&gt; phones &nbsp;&nbsp; 2. Fill in the MAC and select the Vendor &nbsp;&nbsp; 3. If the phone is a Supported phone, when you select the Vendor, the Desc field will auto-populate &nbsp;&nbsp; 4. If the phone does not match any Vendor, choose Other and add your own Desc &nbsp;&nbsp; 5. Click "Add Device" below &nbsp;&nbsp; 6. Now, set the tftp server on the phone to: "s[server_id].trixbox.fonality.com" &nbsp;&nbsp; 7. Reboot your phone &nbsp;&nbsp; 8. Now click on Extensions in your Admin Panel and select the proper extension for this phone (if you haven't created the extension yet, please read How do I add an extension?) &nbsp;&nbsp; 9. Select your newly created device from the list - devices that are already paired with an extension will display the extension in ( ) as pictured:  Supported Remote Phone  &nbsp;&nbsp; 1. Fill in the MAC and Vendor below. &nbsp;&nbsp; 2. Click "Add Device" below. &nbsp;&nbsp; 3. Now, set the tftp server on the phone to: "s[server_id]x.trixbox.fonality.com". Did you notice the x? :) &nbsp;&nbsp; 4. Reboot the phone. &nbsp;&nbsp; 5. If your phone cannot get its configuration from your [server_brand] server then: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. a. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com" &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Set the SIP username and password &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Reboot your phone.  Unsupported Phone (local or remote)  &nbsp;&nbsp; 1. Fill in the MAC and Vendor below. &nbsp;&nbsp; 2. Click "Add Device" below. &nbsp;&nbsp; 3. If LOCAL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id].trixbox.fonality.com" &nbsp;&nbsp; 4. If REMOTE: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Set the "Proxy Server", "Registrar Server" and/or "Domain" options in the phone to "s[server_id]x.trixbox.fonality.com" &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Note the x! &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Set the SIP username and password &nbsp;&nbsp; 5. Reboot your phone. ]]></contentstext>
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<subject><![CDATA[How do I add a phone/device to trixbox PRO?]]></subject>
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<contents><![CDATA[IP-PBX - side:<br /><br />In order to register a remote IP phone with an IP-PBX , you must forward the following ports.&nbsp; Inbound traffic on these ports from the Internet must be forwarded to the internal private IP address of your IP-PBX:<br /><br />&nbsp;&nbsp;&nbsp; * 5060 (UDP).&nbsp; SIP signaling and call setup/teardown.<br />&nbsp;&nbsp;&nbsp; * 10000-20000 (UDP).&nbsp; RTP voice traffic.&nbsp; A voice conversation picks a random pair of ports in that range, so the entire range must be forwarded.<br /><br />Disable SIP handling features on your firewall or router.&nbsp; They will interfere with VoIP traffic.&nbsp; Different manufacturers use different terms:<br /><br />&nbsp;&nbsp;&nbsp; * SIP ALG / Application Layer Gateway.&nbsp; (Seen on Linksys)<br />&nbsp;&nbsp;&nbsp; * SIP Fixup.&nbsp; (Cisco)<br />&nbsp;&nbsp;&nbsp; * SIP Markup.<br />&nbsp;&nbsp;&nbsp; * SIP Session Helper.<br />&nbsp;&nbsp;&nbsp; * SIP Translation.<br /><br />Turn off any of the above.<br /><br />Remote user's side:<br /><br />In addition to above, each remote user also needs to open the following ports on their (remote-side) firewall:<br /><br />&nbsp;&nbsp;&nbsp; * 5060 (UDP), responsible for call setup and teardown.<br />&nbsp;&nbsp;&nbsp; * 10000-20000 (UDP), which are RTP ports used for the transmission of audio.<br />]]></contents>
<contentstext><![CDATA[IP-PBX - side:  In order to register a remote IP phone with an IP-PBX , you must forward the following ports.&nbsp; Inbound traffic on these ports from the Internet must be forwarded to the internal private IP address of your IP-PBX:  &nbsp;&nbsp;&nbsp; * 5060 (UDP).&nbsp; SIP signaling and call setup/teardown. &nbsp;&nbsp;&nbsp; * 10000-20000 (UDP).&nbsp; RTP voice traffic.&nbsp; A voice conversation picks a random pair of ports in that range, so the entire range must be forwarded.  Disable SIP handling features on your firewall or router.&nbsp; They will interfere with VoIP traffic.&nbsp; Different manufacturers use different terms:  &nbsp;&nbsp;&nbsp; * SIP ALG / Application Layer Gateway.&nbsp; (Seen on Linksys) &nbsp;&nbsp;&nbsp; * SIP Fixup.&nbsp; (Cisco) &nbsp;&nbsp;&nbsp; * SIP Markup. &nbsp;&nbsp;&nbsp; * SIP Session Helper. &nbsp;&nbsp;&nbsp; * SIP Translation.  Turn off any of the above.  Remote user's side:  In addition to above, each remote user also needs to open the following ports on their (remote-side) firewall:  &nbsp;&nbsp;&nbsp; * 5060 (UDP), responsible for call setup and teardown. &nbsp;&nbsp;&nbsp; * 10000-20000 (UDP), which are RTP ports used for the transmission of audio. ]]></contentstext>
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<subject><![CDATA[What changes to I need to make to my router for remote phones?]]></subject>
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<contents><![CDATA[If you can match the impedance setting expected by your country's PSTN, you will have much better results in detecting the proper state of your PSTN line as well as have better audio quality.<br /><br />The following settings apply to various countries:<br /><br />&nbsp;&nbsp;&nbsp; * 600 ohms (US/Canada)<br />&nbsp;&nbsp;&nbsp; * 900 ohms (US/Canada)<br />&nbsp;&nbsp;&nbsp; * 220+820//115nF (Germany)<br />&nbsp;&nbsp;&nbsp; * 200+680//100nF (China)<br />&nbsp;&nbsp;&nbsp; * 370+620//310nF (UK)<br />&nbsp;&nbsp;&nbsp; * 180+910//150nF (France)<br />&nbsp;&nbsp;&nbsp; * 270+750//150nF (Sweden)<br />&nbsp;&nbsp;&nbsp; * 120+820//110nF (Norway)<br />&nbsp;&nbsp;&nbsp; * 150+830//72nF (Belgium)<br />&nbsp;&nbsp;&nbsp; * 400+500//330nF (Denmark)<br />]]></contents>
<contentstext><![CDATA[If you can match the impedance setting expected by your country's PSTN, you will have much better results in detecting the proper state of your PSTN line as well as have better audio quality.  The following settings apply to various countries:  &nbsp;&nbsp;&nbsp; * 600 ohms (US/Canada) &nbsp;&nbsp;&nbsp; * 900 ohms (US/Canada) &nbsp;&nbsp;&nbsp; * 220+820//115nF (Germany) &nbsp;&nbsp;&nbsp; * 200+680//100nF (China) &nbsp;&nbsp;&nbsp; * 370+620//310nF (UK) &nbsp;&nbsp;&nbsp; * 180+910//150nF (France) &nbsp;&nbsp;&nbsp; * 270+750//150nF (Sweden) &nbsp;&nbsp;&nbsp; * 120+820//110nF (Norway) &nbsp;&nbsp;&nbsp; * 150+830//72nF (Belgium) &nbsp;&nbsp;&nbsp; * 400+500//330nF (Denmark) ]]></contentstext>
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<subject><![CDATA[What impedence setting should I choose for my SPA3102 FXO port?]]></subject>
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<contents><![CDATA[&nbsp;&nbsp; 1. 1. Provisional 1xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 1.1. 100 Trying<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 1.2. 180 Ringing<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 1.3. 181 Call Is Being Forwarded<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 1.4. 182 Queued<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 1.5. 183 Session Progress<br />&nbsp;&nbsp; 2. 2. Successful 2xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 2.1. 200 OK<br />&nbsp;&nbsp; 3. 3. Redirection 3xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 3.1. 300 Multiple Choices<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 3.2. 301 Moved Permanently<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 3.3. 302 Moved Temporarily<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 3.4. 305 Use Proxy<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 3.5. 380 Alternative Service<br />&nbsp;&nbsp; 4. 4. Request Failure 4xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 4.1. 400 Bad Request<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 4.2. 401 Unauthorized<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 4.3. 402 Payment Required<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 4.4. 403 Forbidden<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 4.5. 404 Not Found<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. 4.6. 405 Method Not Allowed<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. 4.7. 406 Not Acceptable<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8. 4.8. 407 Proxy Authentication Required<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 9. 4.9. 408 Request Timeout<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10. 4.10. 410 Gone<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 11. 4.11. 413 Request Entity Too Large<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 12. 4.12. 414 Request-URI Too Long<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 13. 4.13. 415 Unsupported Media Type<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 14. 4.14. 416 Unsupported URI Scheme<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 15. 4.15. 420 Bad Extension<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 16. 4.16. 421 Extension Required<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 17. 4.17. 423 Interval Too Brief<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 18. 4.18. 480 Temporarily Unavailable<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 19. 4.19. 481 Call/Transaction Does Not Exist<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 20. 4.20. 482 Loop Detected<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 21. 4.21. 483 Too Many Hops<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 22. 4.22. 484 Address Incomplete<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 23. 4.23. 485 Ambiguous<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 24. 4.24. 486 Busy Here<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 25. 4.25. 487 Request Terminated<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 26. 4.26. 488 Not Acceptable Here<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 27. 4.27. 491 Request Pending<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 28. 4.28. 493 Undecipherable<br />&nbsp;&nbsp; 5. 5. Server Failure 5xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 5.1. 500 Server Internal Error<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 5.2. 501 Not Implemented<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 5.3. 502 Bad Gateway<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 5.4. 503 Service Unavailable<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 5.5. 504 Server Time-out<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. 5.6. 505 Version Not Supported<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. 5.7. 513 Message Too Large<br />&nbsp;&nbsp; 6. 6. Global Failures 6xx<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 6.1. 600 Busy Everywhere<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 6.2. 603 Decline<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 6.3. 604 Does Not Exist Anywhere<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 6.4. 606 Not Acceptable]]></contents>
<contentstext><![CDATA[&nbsp;&nbsp; 1. 1. Provisional 1xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 1.1. 100 Trying &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 1.2. 180 Ringing &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 1.3. 181 Call Is Being Forwarded &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 1.4. 182 Queued &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 1.5. 183 Session Progress &nbsp;&nbsp; 2. 2. Successful 2xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 2.1. 200 OK &nbsp;&nbsp; 3. 3. Redirection 3xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 3.1. 300 Multiple Choices &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 3.2. 301 Moved Permanently &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 3.3. 302 Moved Temporarily &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 3.4. 305 Use Proxy &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 3.5. 380 Alternative Service &nbsp;&nbsp; 4. 4. Request Failure 4xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 4.1. 400 Bad Request &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 4.2. 401 Unauthorized &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 4.3. 402 Payment Required &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 4.4. 403 Forbidden &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 4.5. 404 Not Found &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. 4.6. 405 Method Not Allowed &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. 4.7. 406 Not Acceptable &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8. 4.8. 407 Proxy Authentication Required &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 9. 4.9. 408 Request Timeout &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10. 4.10. 410 Gone &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 11. 4.11. 413 Request Entity Too Large &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 12. 4.12. 414 Request-URI Too Long &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 13. 4.13. 415 Unsupported Media Type &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 14. 4.14. 416 Unsupported URI Scheme &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 15. 4.15. 420 Bad Extension &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 16. 4.16. 421 Extension Required &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 17. 4.17. 423 Interval Too Brief &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 18. 4.18. 480 Temporarily Unavailable &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 19. 4.19. 481 Call/Transaction Does Not Exist &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 20. 4.20. 482 Loop Detected &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 21. 4.21. 483 Too Many Hops &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 22. 4.22. 484 Address Incomplete &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 23. 4.23. 485 Ambiguous &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 24. 4.24. 486 Busy Here &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 25. 4.25. 487 Request Terminated &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 26. 4.26. 488 Not Acceptable Here &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 27. 4.27. 491 Request Pending &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 28. 4.28. 493 Undecipherable &nbsp;&nbsp; 5. 5. Server Failure 5xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 5.1. 500 Server Internal Error &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 5.2. 501 Not Implemented &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 5.3. 502 Bad Gateway &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 5.4. 503 Service Unavailable &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. 5.5. 504 Server Time-out &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. 5.6. 505 Version Not Supported &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. 5.7. 513 Message Too Large &nbsp;&nbsp; 6. 6. Global Failures 6xx &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. 6.1. 600 Busy Everywhere &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. 6.2. 603 Decline &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. 6.3. 604 Does Not Exist Anywhere &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. 6.4. 606 Not Acceptable]]></contentstext>
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<subject><![CDATA[Sip Response Codes]]></subject>
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<contents><![CDATA[The Aastra 6730i and 6731i model IP phones come with 8 programmable keys for customization, which are located at the top of the phone.&nbsp; These keys can be set as BLFs (Busy Lamp Fields), Speed Dials, and XML application commands amongst other settings.&nbsp; But as anyone who has used these phone know, out of those keys, two come from factory locked, with pre-configured settings.&nbsp; The number 5 and 6 keys are set as &ldquo;Save&rdquo; and &ldquo;Delete&rdquo; by default, which can be used in conjunction with the XML phone directory.&nbsp; But for those of us who either don&rsquo;t use XML, or wish to be able to configure the phone to their needs, we have to edit the phone&rsquo;s config to unlock those keys for editing.<br /><br />To do this, you must edit your phone&rsquo;s config file to contain the following:<br /><br />prgkey1 locked: 0<br />prgkey2 locked: 0<br /><br />Setting these fields to &lsquo;0&prime; unlocks the buttons from their pre-config, and&nbsp; a &lsquo;1&prime; will return them to a locked state.&nbsp; So if you edit your config to &lsquo;0&prime;, and then load it into the phone using a TFTP Server, you will then gain access to those keys through your phone&rsquo;s web GUI.]]></contents>
<contentstext><![CDATA[The Aastra 6730i and 6731i model IP phones come with 8 programmable keys for customization, which are located at the top of the phone.&nbsp; These keys can be set as BLFs (Busy Lamp Fields), Speed Dials, and XML application commands amongst other settings.&nbsp; But as anyone who has used these phone know, out of those keys, two come from factory locked, with pre-configured settings.&nbsp; The number 5 and 6 keys are set as &ldquo;Save&rdquo; and &ldquo;Delete&rdquo; by default, which can be used in conjunction with the XML phone directory.&nbsp; But for those of us who either don&rsquo;t use XML, or wish to be able to configure the phone to their needs, we have to edit the phone&rsquo;s config to unlock those keys for editing.  To do this, you must edit your phone&rsquo;s config file to contain the following:  prgkey1 locked: 0 prgkey2 locked: 0  Setting these fields to &lsquo;0&prime; unlocks the buttons from their pre-config, and&nbsp; a &lsquo;1&prime; will return them to a locked state.&nbsp; So if you edit your config to &lsquo;0&prime;, and then load it into the phone using a TFTP Server, you will then gain access to those keys through your phone&rsquo;s web GUI.]]></contentstext>
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<subject><![CDATA[How to unlock Aastra 6730i/6731i programmable keys]]></subject>
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<contents><![CDATA[The process for installing trixbox CE on an appliance is very similar to the normal install process.<br /><br />1.) Download the trixbox CE ISO (http://www.trixbox.org/downloads)<br /><br />2.) Burn the ISO image to a CD.&nbsp; Insert the CD into an optical drive that is connected to your appliance.<br /><br />3.) Boot the appliance from the CD drive.&nbsp; This will bring you to the trixbox CE installation screen.&nbsp; <br /><br />4.) Normally at this point, you would use the "install" command.&nbsp; But due to the Appliance needing specific drivers for the hardware and LCD screen, you will issue the "appliance" command.&nbsp; <br /><br />5.) The install process will begin.&nbsp; This is an entirely scripted process, and does not require any further steps.&nbsp; It will take between five and ten minutes to install.<br /><br />The entire process is the same as the normal CE install, with the sole difference being the "appliance" command.&nbsp; <br />]]></contents>
<contentstext><![CDATA[The process for installing trixbox CE on an appliance is very similar to the normal install process.  1.) Download the trixbox CE ISO (http://www.trixbox.org/downloads)  2.) Burn the ISO image to a CD.&nbsp; Insert the CD into an optical drive that is connected to your appliance.  3.) Boot the appliance from the CD drive.&nbsp; This will bring you to the trixbox CE installation screen.&nbsp;   4.) Normally at this point, you would use the "install" command.&nbsp; But due to the Appliance needing specific drivers for the hardware and LCD screen, you will issue the "appliance" command.&nbsp;   5.) The install process will begin.&nbsp; This is an entirely scripted process, and does not require any further steps.&nbsp; It will take between five and ten minutes to install.  The entire process is the same as the normal CE install, with the sole difference being the "appliance" command.&nbsp;  ]]></contentstext>
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<contents><![CDATA[Reset Local Config to factory Default<br />--------------------------------<br />Step 1: Press the 'Menu' key.<br />Step 2: Select 'Settings'.<br />Step 3: Select 'Advanced'.<br />Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.<br />Step 5: Select 'Admin Settings'.<br />Step 6: Select 'Reset to Default'.<br />Step 7: Select 'Reset Local Config'.<br /><br /><br />Reset Device Settings to Factory Default<br />-----------------------------------<br />IP 30x: Hold down the 4, 6, 8 and * keys simultaneously.<br />IP 430: Hold down the 1, 3, 5 and 7 keys simultaneously.<br />IP 50x: Hold down the 4, 6, 8 and * keys simultaneously.<br />IP 60x: Hold down the 4, 6, 8 and * keys simultaneously.<br /><br /><br />Format File System for a Fresh Firmware Load<br />-----------------------------------<br />Step 1: Press the 'Menu' hard key.<br />Step 2: Select 'Settings'.<br />Step 3: Select 'Advanced'.<br />Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key.<br />Step 5: Select 'Admin Settings'.<br />Step 6: Select 'Reset to Default'.<br />Step 7: Select 'Format File System'. <br /><br />NOTE: This method will require the Administrator Password(456)<br />]]></contents>
<contentstext><![CDATA[Reset Local Config to factory Default -------------------------------- Step 1: Press the 'Menu' key. Step 2: Select 'Settings'. Step 3: Select 'Advanced'. Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key. Step 5: Select 'Admin Settings'. Step 6: Select 'Reset to Default'. Step 7: Select 'Reset Local Config'.   Reset Device Settings to Factory Default ----------------------------------- IP 30x: Hold down the 4, 6, 8 and * keys simultaneously. IP 430: Hold down the 1, 3, 5 and 7 keys simultaneously. IP 50x: Hold down the 4, 6, 8 and * keys simultaneously. IP 60x: Hold down the 4, 6, 8 and * keys simultaneously.   Format File System for a Fresh Firmware Load ----------------------------------- Step 1: Press the 'Menu' hard key. Step 2: Select 'Settings'. Step 3: Select 'Advanced'. Step 4: Enter in your admin password to the phone (456). Hit the Enter soft key. Step 5: Select 'Admin Settings'. Step 6: Select 'Reset to Default'. Step 7: Select 'Format File System'.   NOTE: This method will require the Administrator Password(456) ]]></contentstext>
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<subject><![CDATA[How to Restore Factory Defaults/Format a Polycom Phone]]></subject>
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<contents><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to<br />the email sent to the registered user on the Cogentvoip.com account with the account info.<br />This email is sent after you register and order your plan(s) and contains the account information you need to complete the setup process.<br /><br />1. Download the talkswitchTM Start Guide from:<br />http://talkswitch.com/support/phone_system_vs/documentation.aspx<br /><br />Follow the steps to install the system.<br /><br />2. With the talkswitchTM configuration software already installed, open the software<br />and connect to your talkswitchTM system. Select &lsquo;System Information&rsquo; and then &lsquo;IP<br />Configuration&rsquo;.<br /><br />3. Under &lsquo;Set IP Settings&rsquo;: If the talkswitchTM is getting its IP and DNS information from another system set&lsquo;Obtain IP and DNS information automatically&rsquo;. Otherwise set &lsquo;Use the<br />following IP and DNS information&rsquo; and leave the preferred and alternative DNS<br />settings empty.<br /><br />In the &lsquo;Public WAN IP Address&rsquo; box select &lsquo;static&rsquo; and enter your public IP<br />address provided by you Internet Service Provider (Cogentvoip.com preferred<br />configuration). If your public IP address is dynamic (changes), select &lsquo;dynamic&rsquo;<br />and complete the other fields as appropriate for you system.<br /><br />4. Select &lsquo;System Information&rsquo; and then select &lsquo;VoIP Configuration&rsquo;, and finally select the <br /><br />&bull; In the &lsquo;Service Provider Name&rsquo; field, enter Cogentvoip.com.<br />&bull; In the &lsquo;Proxy Server Name&rsquo; field, enter Cogentvoip.com.<br />&bull; In the &lsquo;Registrar Server Name&rsquo; field, enter Cogentvoip.com.<br />&bull; In the &lsquo;Outbound Proxy&rsquo; field, enter Cogentvoip.com.<br />&bull; In the &lsquo;Realm/Domain&rsquo; field, enter Cogentvoip.com.<br /><br />5. In the &lsquo;VoIP Lines&rsquo; fields you can enter the number of simultaneous calls you can<br />support.<br />&bull; If using Cogentvoip.com&rsquo;s Business Unlimited plan these fields should be set to the<br />number of plans you have (e.g. 3 Business Unlimited then set the parameters<br />to 3).<br />&bull; If you subscribe to Cogentvoip.com&rsquo;s Business Office or Business Pro then the<br />number of simultaneous calls is unlimited within the capacity of your Internet<br />access bandwidth. You will require 160 kbps per call if your codec&rsquo;s are<br />configured for G.711 and 48 kbps per cal if your codec&rsquo;s are configured for<br />G.729. Please take into account the usage of data applications if you are<br />sharing the Internet access with you data applications.<br /><br />6. Select &lsquo;System Information&rsquo; and then select &lsquo;VoIP Numbers&rsquo;. Select a VoIP<br />Number tab and check the &lsquo;Activate VoIP Number&rsquo; box, then click on the &lsquo;Service<br />Provider Profile&rsquo; button as shown in Figure 3 below.<br />&bull; Enter you phone number in the &lsquo;Phone Number&rsquo; box.<br />&bull; From the Cogentvoip.com Account Setup email, obtain the &lsquo;Authentication Username&rsquo;<br />and enter it in to the &lsquo;User/Account&rsquo; field. Please ensure you have no hidden<br />characters or trailing spaces.<br />&bull; From the Cogentvoip.com Account Setup email, obtain the &lsquo;Authentication Password&rsquo;<br />and enter it in to the &lsquo;Password&rsquo; field. Please ensure you have no hidden<br />characters or trailing spaces.<br />NOTE: If you have more than one Cogentvoip.com number, select additional VoIP number<br />tab(s) and repeat step 11 for each number.<br /><br />7. Select &lsquo;System Information&rsquo; and the select &lsquo;Line Hunt Groups&rsquo;. Select a Hunt<br />Group to use for accessing the Cogentvoip.com service and ensure Line Type is set to<br />Cogentvoip.com VoIP Numbers. In Figure 4 below, Hunt Group 88 is configured to access all<br />available VoIP lines. For convenience, you can assign a name to the selected Line Hunt<br />Group (e.g. Cogentvoip.com).<br /><br />8. This step is Optional. The talkswitchTM is set to G.729 codec by default which<br />requires 48 kbps per call. This maximizes the number of simultaneous calls, and<br />provides basic audio quality. Alternatively you can select the G.711 codec, which<br />maximizes audio quality and supports Fax, but may limit the number of simultaneous<br />VoIP calls as it requires 160 kbps per call.<br />&bull; To select G.711, select &lsquo;System Information&rsquo; and then select &lsquo;VoIP<br />Configuration&rsquo; then select the Service Provider 1 Profile tab, then select the<br />&lsquo;Codec Options&rsquo; button. In the &lsquo;Codec Options&rsquo; dialog window, use the menu<br />to set the &lsquo;Preferred Codec&rsquo; to G.711.<br /><br />9.) Save settings to by selecting &lsquo;File&rsquo; from the menu and then &lsquo;Save to TalkSwitch&rsquo;.<br />10.) On your firewall open the following ports:<br />5060 UDP Signalling<br />&bull; 6000 &ndash; 6006 UDP (Map to IP address of TalkSwitch Unit ID 1)<br />&bull; 6010 &ndash; 6016 UDP (Map to IP address of TalkSwitch Unit ID 2)<br />&bull; 6020 &ndash; 6026 UDP (Map to IP address of TalkSwitch Unit ID 3)<br />&bull; 6030 &ndash; 6036 UDP (Map to IP address of TalkSwitch Unit ID 4)<br /><br />11.) Confirm that your VoIP Number(s) have registered. Select &lsquo;System Information&rsquo; then<br />select &lsquo;VoIP Numbers&rsquo; and select a Cogentvoip.com number tab. At the bottom of the page,<br />click on the &lsquo;View Registration Status&rsquo;. The &lsquo;Registration Status&rsquo; window will open,<br />showing the registration status for each Cogentvoip.com number.<br />If the status shows a &ldquo;Not Registered&rdquo; result then there is a problem with the<br />configuration or connection to the network. If so, double-check that all information<br />entered in the steps above is correct.<br />]]></contents>
<contentstext><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to the email sent to the registered user on the Cogentvoip.com account with the account info. This email is sent after you register and order your plan(s) and contains the account information you need to complete the setup process.  1. Download the talkswitchTM Start Guide from: http://talkswitch.com/support/phone_system_vs/documentation.aspx  Follow the steps to install the system.  2. With the talkswitchTM configuration software already installed, open the software and connect to your talkswitchTM system. Select &lsquo;System Information&rsquo; and then &lsquo;IP Configuration&rsquo;.  3. Under &lsquo;Set IP Settings&rsquo;: If the talkswitchTM is getting its IP and DNS information from another system set&lsquo;Obtain IP and DNS information automatically&rsquo;. Otherwise set &lsquo;Use the following IP and DNS information&rsquo; and leave the preferred and alternative DNS settings empty.  In the &lsquo;Public WAN IP Address&rsquo; box select &lsquo;static&rsquo; and enter your public IP address provided by you Internet Service Provider (Cogentvoip.com preferred configuration). If your public IP address is dynamic (changes), select &lsquo;dynamic&rsquo; and complete the other fields as appropriate for you system.  4. Select &lsquo;System Information&rsquo; and then select &lsquo;VoIP Configuration&rsquo;, and finally select the   &bull; In the &lsquo;Service Provider Name&rsquo; field, enter Cogentvoip.com. &bull; In the &lsquo;Proxy Server Name&rsquo; field, enter Cogentvoip.com. &bull; In the &lsquo;Registrar Server Name&rsquo; field, enter Cogentvoip.com. &bull; In the &lsquo;Outbound Proxy&rsquo; field, enter Cogentvoip.com. &bull; In the &lsquo;Realm/Domain&rsquo; field, enter Cogentvoip.com.  5. In the &lsquo;VoIP Lines&rsquo; fields you can enter the number of simultaneous calls you can support. &bull; If using Cogentvoip.com&rsquo;s Business Unlimited plan these fields should be set to the number of plans you have (e.g. 3 Business Unlimited then set the parameters to 3). &bull; If you subscribe to Cogentvoip.com&rsquo;s Business Office or Business Pro then the number of simultaneous calls is unlimited within the capacity of your Internet access bandwidth. You will require 160 kbps per call if your codec&rsquo;s are configured for G.711 and 48 kbps per cal if your codec&rsquo;s are configured for G.729. Please take into account the usage of data applications if you are sharing the Internet access with you data applications.  6. Select &lsquo;System Information&rsquo; and then select &lsquo;VoIP Numbers&rsquo;. Select a VoIP Number tab and check the &lsquo;Activate VoIP Number&rsquo; box, then click on the &lsquo;Service Provider Profile&rsquo; button as shown in Figure 3 below. &bull; Enter you phone number in the &lsquo;Phone Number&rsquo; box. &bull; From the Cogentvoip.com Account Setup email, obtain the &lsquo;Authentication Username&rsquo; and enter it in to the &lsquo;User/Account&rsquo; field. Please ensure you have no hidden characters or trailing spaces. &bull; From the Cogentvoip.com Account Setup email, obtain the &lsquo;Authentication Password&rsquo; and enter it in to the &lsquo;Password&rsquo; field. Please ensure you have no hidden characters or trailing spaces. NOTE: If you have more than one Cogentvoip.com number, select additional VoIP number tab(s) and repeat step 11 for each number.  7. Select &lsquo;System Information&rsquo; and the select &lsquo;Line Hunt Groups&rsquo;. Select a Hunt Group to use for accessing the Cogentvoip.com service and ensure Line Type is set to Cogentvoip.com VoIP Numbers. In Figure 4 below, Hunt Group 88 is configured to access all available VoIP lines. For convenience, you can assign a name to the selected Line Hunt Group (e.g. Cogentvoip.com).  8. This step is Optional. The talkswitchTM is set to G.729 codec by default which requires 48 kbps per call. This maximizes the number of simultaneous calls, and provides basic audio quality. Alternatively you can select the G.711 codec, which maximizes audio quality and supports Fax, but may limit the number of simultaneous VoIP calls as it requires 160 kbps per call. &bull; To select G.711, select &lsquo;System Information&rsquo; and then select &lsquo;VoIP Configuration&rsquo; then select the Service Provider 1 Profile tab, then select the &lsquo;Codec Options&rsquo; button. In the &lsquo;Codec Options&rsquo; dialog window, use the menu to set the &lsquo;Preferred Codec&rsquo; to G.711.  9.) Save settings to by selecting &lsquo;File&rsquo; from the menu and then &lsquo;Save to TalkSwitch&rsquo;. 10.) On your firewall open the following ports: 5060 UDP Signalling &bull; 6000 &ndash; 6006 UDP (Map to IP address of TalkSwitch Unit ID 1) &bull; 6010 &ndash; 6016 UDP (Map to IP address of TalkSwitch Unit ID 2) &bull; 6020 &ndash; 6026 UDP (Map to IP address of TalkSwitch Unit ID 3) &bull; 6030 &ndash; 6036 UDP (Map to IP address of TalkSwitch Unit ID 4)  11.) Confirm that your VoIP Number(s) have registered. Select &lsquo;System Information&rsquo; then select &lsquo;VoIP Numbers&rsquo; and select a Cogentvoip.com number tab. At the bottom of the page, click on the &lsquo;View Registration Status&rsquo;. The &lsquo;Registration Status&rsquo; window will open, showing the registration status for each Cogentvoip.com number. If the status shows a &ldquo;Not Registered&rdquo; result then there is a problem with the configuration or connection to the network. If so, double-check that all information entered in the steps above is correct. ]]></contentstext>
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<subject><![CDATA[How to Configure CogentVoip Service with Talkswitch]]></subject>
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<contents><![CDATA[These steps should work for all Aastra Wireless Handsets. <br /><br />Keep in mind, there is a limitation of 4 handsets per base station.<br /><br />During this process, keep the handset within 3 feet of the base station.<br /><br />On the base station:<br /><br />&nbsp;&nbsp; 1. Press Options<br />&nbsp;&nbsp; 2. Select Handset Pairing.&nbsp; (For a 57i CT, Select Preferences, then Handset Pairing)<br />&nbsp;&nbsp; 3. Choose Pair and Press Select.<br /><br />On the Handset:<br /><br />&nbsp;&nbsp; 1. Press the Menu key<br />&nbsp;&nbsp; 2. Select Hset pairing<br />&nbsp;&nbsp; 3. Press the Pair soft key<br /><br /><br />The base station and handset will attempt to connect, and then will display &ldquo;Pairing successful&rdquo; or &ldquo;Pairing failed&rdquo;.<br /><br />If the pairing fails, remove the handset battery. Wait 5 minutes, replace the handset battery and try again. It may take a few tries to pair successfully.&nbsp; If the pairing continues to fail, you may have a bad unit.&nbsp; In this event, you should contact your distributor/reseller for troubleshooting purposes.&nbsp; <br />]]></contents>
<contentstext><![CDATA[These steps should work for all Aastra Wireless Handsets.   Keep in mind, there is a limitation of 4 handsets per base station.  During this process, keep the handset within 3 feet of the base station.  On the base station:  &nbsp;&nbsp; 1. Press Options &nbsp;&nbsp; 2. Select Handset Pairing.&nbsp; (For a 57i CT, Select Preferences, then Handset Pairing) &nbsp;&nbsp; 3. Choose Pair and Press Select.  On the Handset:  &nbsp;&nbsp; 1. Press the Menu key &nbsp;&nbsp; 2. Select Hset pairing &nbsp;&nbsp; 3. Press the Pair soft key   The base station and handset will attempt to connect, and then will display &ldquo;Pairing successful&rdquo; or &ldquo;Pairing failed&rdquo;.  If the pairing fails, remove the handset battery. Wait 5 minutes, replace the handset battery and try again. It may take a few tries to pair successfully.&nbsp; If the pairing continues to fail, you may have a bad unit.&nbsp; In this event, you should contact your distributor/reseller for troubleshooting purposes.&nbsp;  ]]></contentstext>
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<subject><![CDATA[How to Pair an Aastra Handset with a Base Station]]></subject>
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<contents><![CDATA[Phones local to the trixbox Pro use a Network Time Service in order to synchronize with the accurate, local time.&nbsp; This Time Service comes in the form of an IP address given to the phone by the trixbox Pro (If used as a DHCP server) or by your own DHCP server.&nbsp; <br /><br />This technology is called NTP, and though Fonality tries to automate the process of a new phone booting up and correctly synchronizing the time for your location, when the phone is moved from the office to a remote location (including a remote office that does not have a trixbox Pro installed on-site), the automated process will fail.<br /><br />Fonality uses the IP address of your trixbox Pro as the NTP value for every phone.&nbsp; When the phone is moved to a remote location, it stores this IP address.&nbsp; However, the IP address of your server is private and cannot be located from a remote location.&nbsp; Therefore, the time on the phone will begin blinking 12:00 and/or set to some other incorrect time.<br /><br />You can manually configure the NTP settings on your Polycom or Aastra phone if you follow these directions:<br /><br />Aastra<br /><br />&nbsp;&nbsp; 1. Find the IP Address of your phone<br />&nbsp;&nbsp; 2. Open your web browser and navigate to that IP address (type it into the URL field)<br />&nbsp;&nbsp; 3. Login with username admin and password 22222<br />&nbsp;&nbsp; 4. Click on Network Settings<br />&nbsp;&nbsp; 5. Set Time Server 1 to pool.ntp.org<br />&nbsp;&nbsp; 6. Click Save Settings<br />&nbsp;&nbsp; 7. Click on Reset<br />&nbsp;&nbsp; 8. Click on Restart Phone<br /><br />Polycom<br /><br />&nbsp;&nbsp; 1. Find the IP Address of your phone<br />&nbsp;&nbsp; 2. Open your web browser and navigate to that IP address (type it into the URL field)<br />&nbsp;&nbsp; 3. Login with username Polycom and password 456<br />&nbsp;&nbsp; 4. Click on General<br />&nbsp;&nbsp; 5. Set the SNTP Server to pool.ntp.org<br />&nbsp;&nbsp; 6. Click Submit<br /><br />As long as the router in your remote location allows the phone to communicate with pool.ntp.org, the phone should correctly synchronize with the local time.]]></contents>
<contentstext><![CDATA[Phones local to the trixbox Pro use a Network Time Service in order to synchronize with the accurate, local time.&nbsp; This Time Service comes in the form of an IP address given to the phone by the trixbox Pro (If used as a DHCP server) or by your own DHCP server.&nbsp;   This technology is called NTP, and though Fonality tries to automate the process of a new phone booting up and correctly synchronizing the time for your location, when the phone is moved from the office to a remote location (including a remote office that does not have a trixbox Pro installed on-site), the automated process will fail.  Fonality uses the IP address of your trixbox Pro as the NTP value for every phone.&nbsp; When the phone is moved to a remote location, it stores this IP address.&nbsp; However, the IP address of your server is private and cannot be located from a remote location.&nbsp; Therefore, the time on the phone will begin blinking 12:00 and/or set to some other incorrect time.  You can manually configure the NTP settings on your Polycom or Aastra phone if you follow these directions:  Aastra  &nbsp;&nbsp; 1. Find the IP Address of your phone &nbsp;&nbsp; 2. Open your web browser and navigate to that IP address (type it into the URL field) &nbsp;&nbsp; 3. Login with username admin and password 22222 &nbsp;&nbsp; 4. Click on Network Settings &nbsp;&nbsp; 5. Set Time Server 1 to pool.ntp.org &nbsp;&nbsp; 6. Click Save Settings &nbsp;&nbsp; 7. Click on Reset &nbsp;&nbsp; 8. Click on Restart Phone  Polycom  &nbsp;&nbsp; 1. Find the IP Address of your phone &nbsp;&nbsp; 2. Open your web browser and navigate to that IP address (type it into the URL field) &nbsp;&nbsp; 3. Login with username Polycom and password 456 &nbsp;&nbsp; 4. Click on General &nbsp;&nbsp; 5. Set the SNTP Server to pool.ntp.org &nbsp;&nbsp; 6. Click Submit  As long as the router in your remote location allows the phone to communicate with pool.ntp.org, the phone should correctly synchronize with the local time.]]></contentstext>
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<subject><![CDATA[Setting the Time on Polycom and Asstra Phones with Trixbox Pro]]></subject>
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<contents><![CDATA[Polycom phones support speed dial configuration via the phone's interface.<br /><br /><br />Configure Speed Dial Keys<br /><br />&nbsp;&nbsp; 1. Press Menu<br />&nbsp;&nbsp; 2. Press 1 Features<br />&nbsp;&nbsp; 3. Press 1 Contact Directory<br />&nbsp;&nbsp; 4. Press Add<br />&nbsp;&nbsp; 5. Fill out the settings<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Scenario #1 - Speed Dial an Employee<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Fill out the Employees First and Last name in the appropriate fields<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Type in the Employee's extension in the Contact field<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. Leave the Speed Dial Index at the default value (increments automatically)<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. Press Save<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. The configured speed dial will appear as Line 2 (for the first contact) and increment with each new configured speed dial (Line 3, Line 4, etc)<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. Scenario #2 - Speed Dial Park/Voicemail/Other System Extension<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8. For First and Last name - enter whatever text will help you easily identify the speed dial function (e.g. if you want to speed dial Park, type in Park and leave the other<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; name field blank)<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 9. Type in the system extension (Park = 9000, Voicemail = 8555, Group Intercom = *74+Group Extension, etc.).&nbsp; Remember you can only type in ONE extension per <br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; speed dial key<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10. Press Save<br />&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 11. The configured speed dial will appear as Line 2 (for the first key) and increment with each new configured speed dial<br /><br />&nbsp;&nbsp; 6. When you are finished configuring all your speed dial keys, press the Menu button to return to the default screen<br /><br />Executing a Speed Dial<br /><br />&nbsp;&nbsp; 1. Press the button - the phone will dial the extension you configured.<br /><br />Executing a Speed Dial to a System Extension<br /><br />Customers commonly wish to speed dial Park.&nbsp; Once you setup Park as a speed dial on your phone, you can quickly transfer callers to Park via the following sequence:<br /><br />&nbsp;&nbsp; 1. Press the softkey Txfr (may be labeled Trnsfer on some versions of Polycom firmware)<br />&nbsp;&nbsp; 2. Press the Park speed dial key you previously configured<br />&nbsp;&nbsp; 3. Listen for the Parking extension<br />&nbsp;&nbsp; 4. Hangup<br />&nbsp;&nbsp; 5. Inform the destination party of the parked extension<br />&nbsp;&nbsp; 6. That person should dial that extension (e.g. 9001) from their own phone to reconnect with the caller<br />]]></contents>
<contentstext><![CDATA[Polycom phones support speed dial configuration via the phone's interface.   Configure Speed Dial Keys  &nbsp;&nbsp; 1. Press Menu &nbsp;&nbsp; 2. Press 1 Features &nbsp;&nbsp; 3. Press 1 Contact Directory &nbsp;&nbsp; 4. Press Add &nbsp;&nbsp; 5. Fill out the settings &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 1. Scenario #1 - Speed Dial an Employee &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 2. Fill out the Employees First and Last name in the appropriate fields &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3. Type in the Employee's extension in the Contact field &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 4. Leave the Speed Dial Index at the default value (increments automatically) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 5. Press Save &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 6. The configured speed dial will appear as Line 2 (for the first contact) and increment with each new configured speed dial (Line 3, Line 4, etc) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 7. Scenario #2 - Speed Dial Park/Voicemail/Other System Extension &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 8. For First and Last name - enter whatever text will help you easily identify the speed dial function (e.g. if you want to speed dial Park, type in Park and leave the other &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; name field blank) &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 9. Type in the system extension (Park = 9000, Voicemail = 8555, Group Intercom = *74+Group Extension, etc.).&nbsp; Remember you can only type in ONE extension per  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; speed dial key &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 10. Press Save &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 11. The configured speed dial will appear as Line 2 (for the first key) and increment with each new configured speed dial  &nbsp;&nbsp; 6. When you are finished configuring all your speed dial keys, press the Menu button to return to the default screen  Executing a Speed Dial  &nbsp;&nbsp; 1. Press the button - the phone will dial the extension you configured.  Executing a Speed Dial to a System Extension  Customers commonly wish to speed dial Park.&nbsp; Once you setup Park as a speed dial on your phone, you can quickly transfer callers to Park via the following sequence:  &nbsp;&nbsp; 1. Press the softkey Txfr (may be labeled Trnsfer on some versions of Polycom firmware) &nbsp;&nbsp; 2. Press the Park speed dial key you previously configured &nbsp;&nbsp; 3. Listen for the Parking extension &nbsp;&nbsp; 4. Hangup &nbsp;&nbsp; 5. Inform the destination party of the parked extension &nbsp;&nbsp; 6. That person should dial that extension (e.g. 9001) from their own phone to reconnect with the caller ]]></contentstext>
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<subject><![CDATA[How to Program keys on a Polycom IP Phone]]></subject>
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<contents><![CDATA[&nbsp;&nbsp; 1. Ensure that you have created the extension that you wish to use with the phone in your PBX first. Take note of the ID field and the password that you have entered for the extension.<br />&nbsp;&nbsp; 2. Connect the Snom M3 phone to the network and main power supply. The phone will boot up, and providing you have DHCP installed on your network, will obtain an IP and display it at the bottom of the phone&rsquo;s LCD screen.<br />&nbsp;&nbsp; 3. Now point your browser to the web interface of the phone: 'http://&lt;phone-ip-address&gt;'where is the IP address shown on the phone.<br />&nbsp;&nbsp; 4. Enter the password of the phone and click login to access the phone&rsquo;s configuration. For this particular phone, the default username to reach administrator mode is "admin" and the default password is "0000"<br />&nbsp;&nbsp; 5. Click on Identity 1' on the left pane under Setup category so you can configure the phone. (Refer to Figure 1)<br />&nbsp;&nbsp; 6. First make sure that &lsquo;Identity active&rsquo; under Identity 1 is set to On.<br />&nbsp;&nbsp; 7. In the "Display Name" field enter the name you want to be displayed on the Phone display.<br />&nbsp;&nbsp; 8. Now proceed by entering in the "Account" field which is the extension you want to associate with this phone. <br />&nbsp;&nbsp; 9. Enter a password for the account in the &ldquo;Password&rdquo; field. This should match the "Password" you set when creating the extension.<br />&nbsp; 10. In the Registrar field, enter the IP of your PBX server. <br />&nbsp; 11. Repeat the same setting for the Outbound Proxy option.<br />&nbsp; 12. In the "Authentication Username" field enter the authentication ID. <br />&nbsp; 13. Now press "save" to save all changes which is on the bottom of your configuration screen. Your phone will register with the PBX. You can verify that on the Phone System\Line Status page of the 3CX management console.<br />&nbsp; 14. If you want to configure additional lines, please repeat steps 5 to 13 providing that you have additional extensions created on the PBX.]]></contents>
<contentstext><![CDATA[&nbsp;&nbsp; 1. Ensure that you have created the extension that you wish to use with the phone in your PBX first. Take note of the ID field and the password that you have entered for the extension. &nbsp;&nbsp; 2. Connect the Snom M3 phone to the network and main power supply. The phone will boot up, and providing you have DHCP installed on your network, will obtain an IP and display it at the bottom of the phone&rsquo;s LCD screen. &nbsp;&nbsp; 3. Now point your browser to the web interface of the phone: 'http://&lt;phone-ip-address&gt;'where is the IP address shown on the phone. &nbsp;&nbsp; 4. Enter the password of the phone and click login to access the phone&rsquo;s configuration. For this particular phone, the default username to reach administrator mode is "admin" and the default password is "0000" &nbsp;&nbsp; 5. Click on Identity 1' on the left pane under Setup category so you can configure the phone. (Refer to Figure 1) &nbsp;&nbsp; 6. First make sure that &lsquo;Identity active&rsquo; under Identity 1 is set to On. &nbsp;&nbsp; 7. In the "Display Name" field enter the name you want to be displayed on the Phone display. &nbsp;&nbsp; 8. Now proceed by entering in the "Account" field which is the extension you want to associate with this phone.  &nbsp;&nbsp; 9. Enter a password for the account in the &ldquo;Password&rdquo; field. This should match the "Password" you set when creating the extension. &nbsp; 10. In the Registrar field, enter the IP of your PBX server.  &nbsp; 11. Repeat the same setting for the Outbound Proxy option. &nbsp; 12. In the "Authentication Username" field enter the authentication ID.  &nbsp; 13. Now press "save" to save all changes which is on the bottom of your configuration screen. Your phone will register with the PBX. You can verify that on the Phone System\Line Status page of the 3CX management console. &nbsp; 14. If you want to configure additional lines, please repeat steps 5 to 13 providing that you have additional extensions created on the PBX.]]></contentstext>
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<subject><![CDATA[How to Register a Snom M3]]></subject>
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<contents><![CDATA[Some Gateways do not detect a busy tone, dial tone or hangup correctly. This is often caused because the device is pre-configured for use in the USA, and does not understand the call progress tones of your country. In this case you will need to configure the Call Progress Tones for your country.<br /><br />This web site has a database of Call Progress Tones for all Countries<br /><br />http://www.3amsystems.com/wireline/tone-search.htm<br /><br />There is also a link to the exact string that you should enter for Linksys/Sipura devices. Please contact the VOIP Gateway manufacturer for further help on configuring your gateway with the correct Call Progress Tone settings.]]></contents>
<contentstext><![CDATA[Some Gateways do not detect a busy tone, dial tone or hangup correctly. This is often caused because the device is pre-configured for use in the USA, and does not understand the call progress tones of your country. In this case you will need to configure the Call Progress Tones for your country.  This web site has a database of Call Progress Tones for all Countries  http://www.3amsystems.com/wireline/tone-search.htm  There is also a link to the exact string that you should enter for Linksys/Sipura devices. Please contact the VOIP Gateway manufacturer for further help on configuring your gateway with the correct Call Progress Tone settings.]]></contentstext>
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<subject><![CDATA[What Call Progress Tones should I enter for my country?]]></subject>
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<contents><![CDATA[Step 1 &ndash; NAT Settings<br />----------------------------<br />The first stage is to ensure that the NAT (Network Address Translation) configuration is<br />correct in your sip_nat.conf.<br /><br />Login as maint and the password you have set, find the Asterisk link and hover over it.&nbsp; You should see &ldquo;Config Edit&rdquo;. Select and look for &ldquo;sip_nat.conf&rdquo;, enter the following lines in iof which are in quotes;<br /><br />&ldquo;nat=yes&rdquo; <br />-This tells the Asterisk box that some adjustments will need to be made while using SIP protocol over a WAN link.<br /><br />For fixed IP addresses:<br /><br />&ldquo;externip=XXX.XXX.XXX.XXX &ldquo;<br />-This sets the IP address in the SIP packets to your outside IP address. Use the external IP address you have from your ISP.<br /><br />&ldquo;localnet=192.168.X.X/255.255.255.0&rdquo;<br />&nbsp;-This enables the trixbox to know who is registered from the inside LAN and allows the SIP messages to be generated with the correct IP addresses. Use your local subnet settings.<br /><br />Also, under the Security Settings tab, set "Allow Anonymous SIP Calls" to "Yes".<br /><br />Step 2 &ndash; Trunk Settings<br />----------------------------<br />Trunk Name : cogentvoip.com<br />Peer details:<br />dtmf=rfc2833<br />dtmfmode=rfc2833<br />fromuser= local user name [can be used for CLI generation at user level]<br />host=cogentvoip.com<br />insecure=very<br />nat=yes<br />qualify=no<br />secret= [CogentVoip issued proxy password. This is not the web login password]<br />type=peer<br />username=[CogentVoip issued proxy user name. This is not the web login user name]<br /><br /><br />User Context=[local user name]<br />User Details<br />context=from-trunk<br />fromuser= local user name [can be used for CLI generation at user level]<br />host=your External IP address<br />insecure=very<br />nat=yes<br />qualify=no<br />secret=[CogentVoip issued proxy password. This is not the web login password]<br />type=user<br />username=[CogentVoip issued proxy user name. This is not the web login user name]<br /><br /><br />Register String: Username:Password@cogentvoip.com<br /><br />Create your inbound routes for inbound numbers, e.g. 7035790200 routes to phone A,<br />7035790201 routes to phone B and that should cover your CogentVoip trunk setup.<br /><br /><br />Firewall/Router<br />----------------------------<br />You will need to open port 5060 on your firewall to enable the SIP signaling to traverse.<br />You will also need to open the RTP or audio ports. This is different for each customer<br />premise device. Please reference trixbox for this detail.<br />]]></contents>
<contentstext><![CDATA[Step 1 &ndash; NAT Settings ---------------------------- The first stage is to ensure that the NAT (Network Address Translation) configuration is correct in your sip_nat.conf.  Login as maint and the password you have set, find the Asterisk link and hover over it.&nbsp; You should see &ldquo;Config Edit&rdquo;. Select and look for &ldquo;sip_nat.conf&rdquo;, enter the following lines in iof which are in quotes;  &ldquo;nat=yes&rdquo;  -This tells the Asterisk box that some adjustments will need to be made while using SIP protocol over a WAN link.  For fixed IP addresses:  &ldquo;externip=XXX.XXX.XXX.XXX &ldquo; -This sets the IP address in the SIP packets to your outside IP address. Use the external IP address you have from your ISP.  &ldquo;localnet=192.168.X.X/255.255.255.0&rdquo; &nbsp;-This enables the trixbox to know who is registered from the inside LAN and allows the SIP messages to be generated with the correct IP addresses. Use your local subnet settings.  Also, under the Security Settings tab, set "Allow Anonymous SIP Calls" to "Yes".  Step 2 &ndash; Trunk Settings ---------------------------- Trunk Name : cogentvoip.com Peer details: dtmf=rfc2833 dtmfmode=rfc2833 fromuser= local user name [can be used for CLI generation at user level] host=cogentvoip.com insecure=very nat=yes qualify=no secret= [CogentVoip issued proxy password. This is not the web login password] type=peer username=[CogentVoip issued proxy user name. This is not the web login user name]   User Context=[local user name] User Details context=from-trunk fromuser= local user name [can be used for CLI generation at user level] host=your External IP address insecure=very nat=yes qualify=no secret=[CogentVoip issued proxy password. This is not the web login password] type=user username=[CogentVoip issued proxy user name. This is not the web login user name]   Register String: Username:Password@cogentvoip.com  Create your inbound routes for inbound numbers, e.g. 7035790200 routes to phone A, 7035790201 routes to phone B and that should cover your CogentVoip trunk setup.   Firewall/Router ---------------------------- You will need to open port 5060 on your firewall to enable the SIP signaling to traverse. You will also need to open the RTP or audio ports. This is different for each customer premise device. Please reference trixbox for this detail. ]]></contentstext>
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<subject><![CDATA[How to configure trixbox CE  with CogentVoip Service]]></subject>
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<contents><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to the email sent to the registered user on the CogentVoip account with the title &lsquo;CogentVoip Account Setup&rsquo;. This email is sent after you register and contains the account information you need to complete the setup process. From within the 3CX configuration screens:<br />1. Under Lines, click &ldquo;Add VOIP&rdquo;<br />2. Select the radio button labeled select &ldquo;Generic Voip Provider&rdquo; enter &ldquo;CogentVoip&rdquo; as the Name of Provider and click &ldquo;Next&rdquo;.<br />3. Enter &lsquo;CogentVoip&rsquo; in both the Registrar and Outbound Proxy fields.&nbsp; Click on the &lsquo;Source Identification&rsquo; tab, then under &lsquo;SIP Field&rsquo;, select &lsquo;From: Host Part&rsquo;.&nbsp; For the Value drop down, select &lsquo;Custom Field&rsquo;.&nbsp; Set the Custom value field to &lsquo;CogentVoip.com&rsquo;, then select the match any fields option.&nbsp; Click &ldquo;Next&rdquo; again.<br />4. In the External Number field, enter a 1 followed by your DID (11 total digits).<br />5. From the CogentVoip Account Setup email, obtain the &lsquo;Authentication Username&rsquo; and enter it in to the &lsquo;Authentication ID&rsquo; field of the 3CX. Please ensure you have no hidden characters or trailing spaces. <br />6. From the CogentVoip Account Setup email, obtain the &lsquo;Authentication Password&rsquo; and enter it in to the &lsquo;Authentication Password&rsquo; field of the 3CX. Please ensure you have no hidden characters or trailing spaces. Click &ldquo;Next&rdquo;.<br />7. Enter any Outbound Rules you wish to impose on your network. Click &ldquo;Finish&rdquo;.<br />8. Your Account should now be registered; you can view the status of the Voip Trunk by clicking &lsquo;Line Status&rsquo;, which is located under &lsquo;Phone System&rsquo;&nbsp; on the left hand side of the 3CX Web Interface.<br />]]></contents>
<contentstext><![CDATA[Configuring the system is very simple. Before you start please ensure you have access to the email sent to the registered user on the CogentVoip account with the title &lsquo;CogentVoip Account Setup&rsquo;. This email is sent after you register and contains the account information you need to complete the setup process. From within the 3CX configuration screens: 1. Under Lines, click &ldquo;Add VOIP&rdquo; 2. Select the radio button labeled select &ldquo;Generic Voip Provider&rdquo; enter &ldquo;CogentVoip&rdquo; as the Name of Provider and click &ldquo;Next&rdquo;. 3. Enter &lsquo;CogentVoip&rsquo; in both the Registrar and Outbound Proxy fields.&nbsp; Click on the &lsquo;Source Identification&rsquo; tab, then under &lsquo;SIP Field&rsquo;, select &lsquo;From: Host Part&rsquo;.&nbsp; For the Value drop down, select &lsquo;Custom Field&rsquo;.&nbsp; Set the Custom value field to &lsquo;CogentVoip.com&rsquo;, then select the match any fields option.&nbsp; Click &ldquo;Next&rdquo; again. 4. In the External Number field, enter a 1 followed by your DID (11 total digits). 5. From the CogentVoip Account Setup email, obtain the &lsquo;Authentication Username&rsquo; and enter it in to the &lsquo;Authentication ID&rsquo; field of the 3CX. Please ensure you have no hidden characters or trailing spaces.  6. From the CogentVoip Account Setup email, obtain the &lsquo;Authentication Password&rsquo; and enter it in to the &lsquo;Authentication Password&rsquo; field of the 3CX. Please ensure you have no hidden characters or trailing spaces. Click &ldquo;Next&rdquo;. 7. Enter any Outbound Rules you wish to impose on your network. Click &ldquo;Finish&rdquo;. 8. Your Account should now be registered; you can view the status of the Voip Trunk by clicking &lsquo;Line Status&rsquo;, which is located under &lsquo;Phone System&rsquo;&nbsp; on the left hand side of the 3CX Web Interface. ]]></contentstext>
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<subject><![CDATA[How to configure 3CX with CogentVoip Service]]></subject>
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<contents><![CDATA[Here is a complete guide of the Grandstream IVR commands for use with the Handytone Series.<br /><br />Menu Voice Prompt Options<br /><br />To enter this voice prompt menu, press the LED button or press &ldquo;***&rdquo; from the analog phone.<br /><br />Main Menu &ldquo;Enter a Menu Option&rdquo; Press &ldquo;*&rdquo; for the next menu option<br />Press &ldquo;#&rdquo; to return to the main menu<br /><br />Available Menu Options: 01-06, 47, 86, 99.<br /><br />01 &ldquo;DHCP Mode&rdquo;, &ldquo;Static IP Mode&rdquo;<br />Press &ldquo;9&rdquo; to toggle the selection<br /><br />If using &ldquo;Static IP Mode&rdquo;, configure the IP address information using menus 02 to 05.<br />If using &ldquo;Dynamic IP Mode&rdquo;, all IP address information comes from the DHCP server automatically after reboot.<br /><br />02 &ldquo;IP Address &ldquo; + IP address The current WAN IP address is announced<br />If using &ldquo;Static IP Mode&rdquo;, enter 12 digit new IP address.<br /><br />03 &ldquo;Subnet &ldquo; + IP address Same as menu 02<br /><br />04 &ldquo;Gateway &ldquo; + IP address Same as menu 02<br /><br />05 &ldquo;DNS Server &ldquo; + IP address Same as menu 02<br /><br />07 Preferred Vocoder Press &ldquo;9&rdquo; to move to the next selection in the list:<br />&bull; PCM U / PCM A<br />&bull; G.723<br />&bull; G.729<br />&bull; G.726<br />&bull; iLBC<br /><br />12 WAN Port Web Access Press &ldquo;9&rdquo; to toggle between enable / disable<br /><br />13 Firmware Server IP Address<br />Announces current Firmware Server IP address. Enter 12 digit new IP address.<br /><br />14 Configuration Server IP Address Announces current Config Server Path IP address.<br />Enter 12 digit new IP address.<br /><br />15 Upgrade Protocol Upgrade protocol for firmware and configuration update.<br />Press &ldquo;9&rdquo; to toggle between TFTP / HTTP<br /><br />16 Firmware Version Firmware version information.<br /><br />17 Firmware Upgrade Firmware upgrade mode. Press &ldquo;9&rdquo; to toggle among the<br />following three options:<br />- always check<br />- check when pre/suffix changes<br />- never upgrade<br /><br />47 &ldquo;Direct IP Calling&rdquo; Enter a 12 digit IP address to make a direct IP call, after dial tone. (See &ldquo;Make a Direct IP Call&rdquo;.)<br /><br />99 &ldquo;RESET&rdquo; Press &ldquo;9&rdquo; to reboot the device; or<br />Enter encoded MAC address to restore factory default<br />setting.<br /><br />RESET VIA IVR<br />1.) Find the MAC address of the device. It is a 12 digits HEX number located on the bottom<br />of the unit.<br /><br />2.) Encode the MAC address. Please use the following mapping:<br />0-9: 0-9<br />A: 22<br />B: 222<br />C: 2222<br />D: 33<br />E: 333<br />F: 3333<br />For example, if the MAC address is 000b8200e395, it should be encoded as<br />&ldquo;0002228200333395&rdquo;.<br /><br />3.) To perform factory reset:<br />a.) Pick up the headset and dial &ldquo;***&rdquo; for voice prompt.<br />b.) Enter &ldquo;99&rdquo; and get the voice prompt &ldquo;Reset&rdquo;.<br />c.) Enter the encoded MAC address of the device.<br />d.) Wait for 15 seconds. The device will reboot automatically and restore to factory<br />default setting.<br /><br />Any &ldquo;Invalid Entry&rdquo; Automatically returns to main menu.<br /><br />NOTE:<br />&bull; Once the button is pressed, you will hear the voice prompt main menu. If the button is pressed<br />again, while it is already in the voice prompt menu, it jumps to &ldquo;Direct IP Call&rdquo; option and a dial<br />tone is prompted<br />&bull; &ldquo;*&rdquo; shifts down to the next menu option<br />&bull; &ldquo;#&rdquo; returns to the main menu<br />&bull; &ldquo;9&rdquo; functions as the ENTER key in many cases to confirm an option<br />&bull; All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP<br />address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26<br />should be key in like 192168000026, no dot needed while input). Once all of the digits are<br />collected, the input will be processed.<br />&bull; Key entry can not be deleted but the phone may prompt error once it is detected<br />]]></contents>
<contentstext><![CDATA[Here is a complete guide of the Grandstream IVR commands for use with the Handytone Series.  Menu Voice Prompt Options  To enter this voice prompt menu, press the LED button or press &ldquo;***&rdquo; from the analog phone.  Main Menu &ldquo;Enter a Menu Option&rdquo; Press &ldquo;*&rdquo; for the next menu option Press &ldquo;#&rdquo; to return to the main menu  Available Menu Options: 01-06, 47, 86, 99.  01 &ldquo;DHCP Mode&rdquo;, &ldquo;Static IP Mode&rdquo; Press &ldquo;9&rdquo; to toggle the selection  If using &ldquo;Static IP Mode&rdquo;, configure the IP address information using menus 02 to 05. If using &ldquo;Dynamic IP Mode&rdquo;, all IP address information comes from the DHCP server automatically after reboot.  02 &ldquo;IP Address &ldquo; + IP address The current WAN IP address is announced If using &ldquo;Static IP Mode&rdquo;, enter 12 digit new IP address.  03 &ldquo;Subnet &ldquo; + IP address Same as menu 02  04 &ldquo;Gateway &ldquo; + IP address Same as menu 02  05 &ldquo;DNS Server &ldquo; + IP address Same as menu 02  07 Preferred Vocoder Press &ldquo;9&rdquo; to move to the next selection in the list: &bull; PCM U / PCM A &bull; G.723 &bull; G.729 &bull; G.726 &bull; iLBC  12 WAN Port Web Access Press &ldquo;9&rdquo; to toggle between enable / disable  13 Firmware Server IP Address Announces current Firmware Server IP address. Enter 12 digit new IP address.  14 Configuration Server IP Address Announces current Config Server Path IP address. Enter 12 digit new IP address.  15 Upgrade Protocol Upgrade protocol for firmware and configuration update. Press &ldquo;9&rdquo; to toggle between TFTP / HTTP  16 Firmware Version Firmware version information.  17 Firmware Upgrade Firmware upgrade mode. Press &ldquo;9&rdquo; to toggle among the following three options: - always check - check when pre/suffix changes - never upgrade  47 &ldquo;Direct IP Calling&rdquo; Enter a 12 digit IP address to make a direct IP call, after dial tone. (See &ldquo;Make a Direct IP Call&rdquo;.)  99 &ldquo;RESET&rdquo; Press &ldquo;9&rdquo; to reboot the device; or Enter encoded MAC address to restore factory default setting.  RESET VIA IVR 1.) Find the MAC address of the device. It is a 12 digits HEX number located on the bottom of the unit.  2.) Encode the MAC address. Please use the following mapping: 0-9: 0-9 A: 22 B: 222 C: 2222 D: 33 E: 333 F: 3333 For example, if the MAC address is 000b8200e395, it should be encoded as &ldquo;0002228200333395&rdquo;.  3.) To perform factory reset: a.) Pick up the headset and dial &ldquo;***&rdquo; for voice prompt. b.) Enter &ldquo;99&rdquo; and get the voice prompt &ldquo;Reset&rdquo;. c.) Enter the encoded MAC address of the device. d.) Wait for 15 seconds. The device will reboot automatically and restore to factory default setting.  Any &ldquo;Invalid Entry&rdquo; Automatically returns to main menu.  NOTE: &bull; Once the button is pressed, you will hear the voice prompt main menu. If the button is pressed again, while it is already in the voice prompt menu, it jumps to &ldquo;Direct IP Call&rdquo; option and a dial tone is prompted &bull; &ldquo;*&rdquo; shifts down to the next menu option &bull; &ldquo;#&rdquo; returns to the main menu &bull; &ldquo;9&rdquo; functions as the ENTER key in many cases to confirm an option &bull; All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26 should be key in like 192168000026, no dot needed while input). Once all of the digits are collected, the input will be processed. &bull; Key entry can not be deleted but the phone may prompt error once it is detected ]]></contentstext>
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<subject><![CDATA[Grandstream Handytone Interactive Voice Response Menu]]></subject>
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<contents><![CDATA[VoIP : Voice over Internet Protocol / Voice over IP<br />ADSL : Asymmetric digital subscriber line. Type of DSL connection where data flow is greater in one direction than the other.<br />ATA : Analog Telephone Adapter<br />Unit to connect a regular anaog phone to the internet to make and receive VoIP calls. Meaning you can use your existing analog phone (eg Uniden cordless, Telstra corded) and start making VoIP calls. <br />Automated Attendent : Automated receptionist answering calls using a recorded message and directing calls to the required area of the business. <br />Bandwidth : The transmission capacity of a specific device or network.<br />Bridge : A device that connects two different king od local networks. ie wireless network to a wired ethernet network.<br />Broadband : An always on, fast internet connection. Recommended for VoIP<br />CID : Caller ID/Caller ID Display<br />Codec : Coder-decoder<br />Software that converts audio signals to compressed digital signals.<br />DHCP : Cynamic Host Configuration Protocol. A protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers.<br />DMZ : Demilitarized Zone : Removes the router's firewall protection from on PC, allowing it to be "seen" from the internet.<br />DID : Direct in Dial - allows callers to call a specific extension/person and bypass the main reception number/automated attendent.<br />DSL : Digital Subscriber Line. An always on broadband connection over traditional phone lines. <br />Firmware : The embedded software that runs a networking device.<br />Full Duplex : The ability of a networking device to receive and transmit data simultaneously.<br />Gateway : A device that interconnects networks with different, incompatible communications protocols.<br />IP : Internet Protocol.<br />IP Address : The address used to identify a computer or device on a network. (example 192.0.00.0)<br />IPSec : Internet Protocol Security. A VPN protocol used to implement secure exchange of packets at the IP layer. Most widely used for enabling virtual private networks.<br />ITSP : Internet Telephony Service Provider. A company that provides voice communicationg service over the Internet (VoIP provider) ie engin, gotalk, MyNetFone, etc.<br />IVR : Interactive Voice Response<br />LAN : Local Area Network. The computers and networking products that make up the network in your home of office.<br />MAC Address : Media Access Control. The unique address that a manufacturer assigns to each networking device. Typically 12 digits in length.<br />Managed Switch : A network switch with an IP address that lets you monitor and administer your network.<br />NAT : Network Address Translation<br />PBX : Private Branch Exchange : A private telephone network used within a business. Users share a certain number of outside lines to make external phone calls. See Linksys SPA9000, epygi Quadro, Switchvox, Asterisk<br />PoE : Power over Ethernet : A technology enabling and Ethernet network cable to deliver both data and power.<br />POTS : Plain old telephone service<br />PSTN : Public Switch Telephone Network<br />QoS : Quality of Service<br />SIP : Session Initiation Protocol<br />Softphone : Telephony Software&nbsp; installed on a PC/Mac enabling the user to make and receive phone calls over the internet connection. Requires software, PC, microphone and headset/speakers.<br />]]></contents>
<contentstext><![CDATA[VoIP : Voice over Internet Protocol / Voice over IP ADSL : Asymmetric digital subscriber line. Type of DSL connection where data flow is greater in one direction than the other. ATA : Analog Telephone Adapter Unit to connect a regular anaog phone to the internet to make and receive VoIP calls. Meaning you can use your existing analog phone (eg Uniden cordless, Telstra corded) and start making VoIP calls.  Automated Attendent : Automated receptionist answering calls using a recorded message and directing calls to the required area of the business.  Bandwidth : The transmission capacity of a specific device or network. Bridge : A device that connects two different king od local networks. ie wireless network to a wired ethernet network. Broadband : An always on, fast internet connection. Recommended for VoIP CID : Caller ID/Caller ID Display Codec : Coder-decoder Software that converts audio signals to compressed digital signals. DHCP : Cynamic Host Configuration Protocol. A protocol that lets one device on a local network, known as a DHCP server, assign temporary IP addresses to the other network devices, typically computers. DMZ : Demilitarized Zone : Removes the router's firewall protection from on PC, allowing it to be "seen" from the internet. DID : Direct in Dial - allows callers to call a specific extension/person and bypass the main reception number/automated attendent. DSL : Digital Subscriber Line. An always on broadband connection over traditional phone lines.  Firmware : The embedded software that runs a networking device. Full Duplex : The ability of a networking device to receive and transmit data simultaneously. Gateway : A device that interconnects networks with different, incompatible communications protocols. IP : Internet Protocol. IP Address : The address used to identify a computer or device on a network. (example 192.0.00.0) IPSec : Internet Protocol Security. A VPN protocol used to implement secure exchange of packets at the IP layer. Most widely used for enabling virtual private networks. ITSP : Internet Telephony Service Provider. A company that provides voice communicationg service over the Internet (VoIP provider) ie engin, gotalk, MyNetFone, etc. IVR : Interactive Voice Response LAN : Local Area Network. The computers and networking products that make up the network in your home of office. MAC Address : Media Access Control. The unique address that a manufacturer assigns to each networking device. Typically 12 digits in length. Managed Switch : A network switch with an IP address that lets you monitor and administer your network. NAT : Network Address Translation PBX : Private Branch Exchange : A private telephone network used within a business. Users share a certain number of outside lines to make external phone calls. See Linksys SPA9000, epygi Quadro, Switchvox, Asterisk PoE : Power over Ethernet : A technology enabling and Ethernet network cable to deliver both data and power. POTS : Plain old telephone service PSTN : Public Switch Telephone Network QoS : Quality of Service SIP : Session Initiation Protocol Softphone : Telephony Software&nbsp; installed on a PC/Mac enabling the user to make and receive phone calls over the internet connection. Requires software, PC, microphone and headset/speakers. ]]></contentstext>
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<subject><![CDATA[VoIP Glossary : What do the acronyms and VoIP terms mean?]]></subject>
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<contents><![CDATA[1)&nbsp;&nbsp;&nbsp; Power down your Aastra Telephone System.<br />2)&nbsp;&nbsp;&nbsp; Hold down the number 1 and # key(s). <br />3)&nbsp;&nbsp;&nbsp; Power on your Aastra Telephone System.<br />4)&nbsp;&nbsp;&nbsp; Continue to hold down 1 and # key(s) while the system initializes.&nbsp; <br />5)&nbsp;&nbsp;&nbsp; Once the system initializes, you will receive the following message:<br />&ldquo;Application Not Found&rdquo;<br />&ldquo;To restore go to Web Recovery Page&rdquo;<br />&ldquo;Web recovery page can be found at the following IP Address: XXX.XXX.XXX.XXX<br />6)&nbsp;&nbsp;&nbsp; Enter IP Address into your web browser to visit the Web Recovery Interface.<br />6a.&nbsp;&nbsp;&nbsp; The Web Recovery will ask for 2 fields to be filled: File Name and TFTP Server.&nbsp; File Name will be the name of your downloaded Firmware, and the TFTP Server Will be the IP of a PC running a TFTP program like Pumpkin.<br />6b.&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />6c.&nbsp;&nbsp;&nbsp; To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />6d.&nbsp;&nbsp;&nbsp; Browse to the Aastra section and download the Firmware Update for your model of IP telephone.<br />6e.&nbsp;&nbsp;&nbsp; Save and Unzip the File to your local directory<br />6f.&nbsp;&nbsp;&nbsp; Run the Pumpkin Program from its installed directory.<br />6g.&nbsp;&nbsp;&nbsp; Once the Program is open. Click the Options button.&nbsp; This will open an option tab.<br />6h.&nbsp;&nbsp;&nbsp; Set the TFTP File system Root to your Firmware directory.&nbsp; Select the &ldquo;Give files&rdquo; and &ldquo;take files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure the Pumpkin Server is running by verifying in the bottom right corner of&nbsp; the Pumpkin window. <br />6i.&nbsp;&nbsp;&nbsp; Return to your internet browser, which should still be at the Web Recovery Interface.&nbsp;&nbsp; Enter the exact file name of the firmware to be installed in the &ldquo;filename&rdquo; box.&nbsp; Enter the IP address of the Computer running Pumpkin into the TFTP box.&nbsp;&nbsp; <br />6j.&nbsp;&nbsp;&nbsp; Select the download firmware button.&nbsp; Install will take around 30 seconds.&nbsp; <br />7)&nbsp;&nbsp;&nbsp; You have completed the Firmware Installation of an Aastra IP Telephone.<br />]]></contents>
<contentstext><![CDATA[1)&nbsp;&nbsp;&nbsp; Power down your Aastra Telephone System. 2)&nbsp;&nbsp;&nbsp; Hold down the number 1 and # key(s).  3)&nbsp;&nbsp;&nbsp; Power on your Aastra Telephone System. 4)&nbsp;&nbsp;&nbsp; Continue to hold down 1 and # key(s) while the system initializes.&nbsp;  5)&nbsp;&nbsp;&nbsp; Once the system initializes, you will receive the following message: &ldquo;Application Not Found&rdquo; &ldquo;To restore go to Web Recovery Page&rdquo; &ldquo;Web recovery page can be found at the following IP Address: XXX.XXX.XXX.XXX 6)&nbsp;&nbsp;&nbsp; Enter IP Address into your web browser to visit the Web Recovery Interface. 6a.&nbsp;&nbsp;&nbsp; The Web Recovery will ask for 2 fields to be filled: File Name and TFTP Server.&nbsp; File Name will be the name of your downloaded Firmware, and the TFTP Server Will be the IP of a PC running a TFTP program like Pumpkin. 6b.&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe) 6c.&nbsp;&nbsp;&nbsp; To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link. 6d.&nbsp;&nbsp;&nbsp; Browse to the Aastra section and download the Firmware Update for your model of IP telephone. 6e.&nbsp;&nbsp;&nbsp; Save and Unzip the File to your local directory 6f.&nbsp;&nbsp;&nbsp; Run the Pumpkin Program from its installed directory. 6g.&nbsp;&nbsp;&nbsp; Once the Program is open. Click the Options button.&nbsp; This will open an option tab. 6h.&nbsp;&nbsp;&nbsp; Set the TFTP File system Root to your Firmware directory.&nbsp; Select the &ldquo;Give files&rdquo; and &ldquo;take files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure the Pumpkin Server is running by verifying in the bottom right corner of&nbsp; the Pumpkin window.  6i.&nbsp;&nbsp;&nbsp; Return to your internet browser, which should still be at the Web Recovery Interface.&nbsp;&nbsp; Enter the exact file name of the firmware to be installed in the &ldquo;filename&rdquo; box.&nbsp; Enter the IP address of the Computer running Pumpkin into the TFTP box.&nbsp;&nbsp;  6j.&nbsp;&nbsp;&nbsp; Select the download firmware button.&nbsp; Install will take around 30 seconds.&nbsp;  7)&nbsp;&nbsp;&nbsp; You have completed the Firmware Installation of an Aastra IP Telephone. ]]></contentstext>
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<subject><![CDATA[How to Install Firmware on an Aastra Phone]]></subject>
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<contents><![CDATA[1)&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />2)&nbsp;&nbsp;&nbsp; To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />2a.&nbsp;&nbsp;&nbsp; Browse to the Aastra section and download the Firmware Update for your model of IP telephone.<br />2b.&nbsp;&nbsp;&nbsp; Save and Unzip the File to your local directory<br />3)&nbsp;&nbsp;&nbsp; Run the Pumpkin program from its install directory.&nbsp; <br />3a.&nbsp;&nbsp;&nbsp; Once the Program is open, Click the options button.&nbsp; This will open an option tab.<br />3b.&nbsp;&nbsp;&nbsp; Set the TFTP file system to your Firmware directory.&nbsp; Select the &ldquo;Give Files&rdquo; and &ldquo;Take Files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. <br />4)&nbsp;&nbsp;&nbsp; Find your Aastra Telephone&rsquo;s IP address.<br />4a.&nbsp;&nbsp;&nbsp; To find the address, enter the phone&rsquo;s Options menu.&nbsp;&nbsp;&nbsp; Scroll down and select the Network option.<br />4b.&nbsp;&nbsp;&nbsp; When prompted, enter your password. The Default setting for Aastra phones is Password: 22222.<br />4c.&nbsp;&nbsp;&nbsp; Scroll down the network menu to the IP Address option and select it.&nbsp; This will display your telephone&rsquo;s IP address. <br />5)&nbsp;&nbsp;&nbsp; Open an internet browser.&nbsp; Enter your telephone&rsquo;s IP address into the navigation bar. <br />5a.&nbsp;&nbsp;&nbsp; When prompted, enter your username and password. The Default setting for Aastra phones is Username: Admin, Password: 22222. <br />6)&nbsp;&nbsp;&nbsp; This will display your Telephones web interface.<br />7)&nbsp;&nbsp;&nbsp; Select the FIRMWARE UPDATE link on the bottom left side of the screen.<br />8)&nbsp;&nbsp;&nbsp; This will display the Firmware Update interface.&nbsp; <br />8a.&nbsp;&nbsp;&nbsp; In the TFTP Server IP box enter the IP of your Pumpkin Server.<br />8b.&nbsp;&nbsp;&nbsp; In the File Name box enter the file name of the firmware file that is to be loaded (See below image for an example.) <br />8c.&nbsp;&nbsp;&nbsp; Click Download Firmware.&nbsp; This will begin the firmware update, which will take several minutes.&nbsp; Once complete your IP telephone will reboot.&nbsp; Once the telephone returns to its normal user interface, the update is complete. <br />&nbsp;<br /><br /><br /><br /><br />]]></contents>
<contentstext><![CDATA[1)&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe) 2)&nbsp;&nbsp;&nbsp; To Download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link. 2a.&nbsp;&nbsp;&nbsp; Browse to the Aastra section and download the Firmware Update for your model of IP telephone. 2b.&nbsp;&nbsp;&nbsp; Save and Unzip the File to your local directory 3)&nbsp;&nbsp;&nbsp; Run the Pumpkin program from its install directory.&nbsp;  3a.&nbsp;&nbsp;&nbsp; Once the Program is open, Click the options button.&nbsp; This will open an option tab. 3b.&nbsp;&nbsp;&nbsp; Set the TFTP file system to your Firmware directory.&nbsp; Select the &ldquo;Give Files&rdquo; and &ldquo;Take Files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window.  4)&nbsp;&nbsp;&nbsp; Find your Aastra Telephone&rsquo;s IP address. 4a.&nbsp;&nbsp;&nbsp; To find the address, enter the phone&rsquo;s Options menu.&nbsp;&nbsp;&nbsp; Scroll down and select the Network option. 4b.&nbsp;&nbsp;&nbsp; When prompted, enter your password. The Default setting for Aastra phones is Password: 22222. 4c.&nbsp;&nbsp;&nbsp; Scroll down the network menu to the IP Address option and select it.&nbsp; This will display your telephone&rsquo;s IP address.  5)&nbsp;&nbsp;&nbsp; Open an internet browser.&nbsp; Enter your telephone&rsquo;s IP address into the navigation bar.  5a.&nbsp;&nbsp;&nbsp; When prompted, enter your username and password. The Default setting for Aastra phones is Username: Admin, Password: 22222.  6)&nbsp;&nbsp;&nbsp; This will display your Telephones web interface. 7)&nbsp;&nbsp;&nbsp; Select the FIRMWARE UPDATE link on the bottom left side of the screen. 8)&nbsp;&nbsp;&nbsp; This will display the Firmware Update interface.&nbsp;  8a.&nbsp;&nbsp;&nbsp; In the TFTP Server IP box enter the IP of your Pumpkin Server. 8b.&nbsp;&nbsp;&nbsp; In the File Name box enter the file name of the firmware file that is to be loaded (See below image for an example.)  8c.&nbsp;&nbsp;&nbsp; Click Download Firmware.&nbsp; This will begin the firmware update, which will take several minutes.&nbsp; Once complete your IP telephone will reboot.&nbsp; Once the telephone returns to its normal user interface, the update is complete.  &nbsp;     ]]></contentstext>
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<subject><![CDATA[How to Update Firmware on an Aastra Phone]]></subject>
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<contents><![CDATA[1)&nbsp;&nbsp;&nbsp; Power down your Polycom Telephone System.<br />2)&nbsp;&nbsp;&nbsp; Download newest Polycom Firmware and Launch Pumpkin TFTP Program<br />2a.&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe)<br />2b.&nbsp;&nbsp;&nbsp; To download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link.<br />2c.&nbsp;&nbsp;&nbsp; Browse to the Polycom section and download the Firmware Update for your model of IP telephone.&nbsp; Save and Unzip the File to your local firmware directory<br />2d.&nbsp;&nbsp;&nbsp; Run the Pumpkin Program from its installed directory.<br />2e.&nbsp;&nbsp;&nbsp; Once the Program is open. Click the Options button.&nbsp; This will open an option tab.<br />2f.&nbsp;&nbsp;&nbsp; Set the TFTP File system Root to your Firmware directory.&nbsp; Select the &ldquo;Give files&rdquo; and &ldquo;take files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure&nbsp;&nbsp; the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window. <br />3)&nbsp;&nbsp;&nbsp; Power on your Polycom Telephone System.<br />4)&nbsp;&nbsp;&nbsp; As the unit initializes, you will be given the option to Start, go to Setup, or Exit.&nbsp; Chose the setup option on the telephone.<br />a.&nbsp;&nbsp;&nbsp; When prompted enter your administrator password.&nbsp; The default Polycom Password is 456.<br />5)&nbsp;&nbsp;&nbsp; You will then be taken to a setup menu.&nbsp; Scroll down and select the SERVER MENU option.&nbsp; <br />6)&nbsp;&nbsp;&nbsp; Once inside the SERVER MENU, select the SERVER TYPE option and change it to TRIVIAL FTP.&nbsp; Save your choice and scroll down to the SERVER ADDERESS option.&nbsp; Enter the IP of the Computer running your Pumpkin Server.&nbsp; <br />7)&nbsp;&nbsp;&nbsp; To save your changes, continue to select exit until you are prompted to save or discard your changes.&nbsp; Select the SAVE AND REBOOT option.<br />8)&nbsp;&nbsp;&nbsp; Your Polycom Telephone will reboot and begin to update. This will take as long as fifteen to thirty minutes.&nbsp; <br />9)&nbsp;&nbsp;&nbsp; Congratulations, you have completed the loading of firmware on your Polycom Telephone system!<br />]]></contents>
<contentstext><![CDATA[1)&nbsp;&nbsp;&nbsp; Power down your Polycom Telephone System. 2)&nbsp;&nbsp;&nbsp; Download newest Polycom Firmware and Launch Pumpkin TFTP Program 2a.&nbsp;&nbsp;&nbsp; Download and Install Pumpkin (http://kin.klever.net/dist/pumpkin-2.7.2.exe) 2b.&nbsp;&nbsp;&nbsp; To download the newest firmware, visit the 888VoipStore.com support downloads site at www.888Voipstore.com/support/ and following the Downloads link. 2c.&nbsp;&nbsp;&nbsp; Browse to the Polycom section and download the Firmware Update for your model of IP telephone.&nbsp; Save and Unzip the File to your local firmware directory 2d.&nbsp;&nbsp;&nbsp; Run the Pumpkin Program from its installed directory. 2e.&nbsp;&nbsp;&nbsp; Once the Program is open. Click the Options button.&nbsp; This will open an option tab. 2f.&nbsp;&nbsp;&nbsp; Set the TFTP File system Root to your Firmware directory.&nbsp; Select the &ldquo;Give files&rdquo; and &ldquo;take files&rdquo; options.&nbsp; Apply the settings and select &ldquo;OK&rdquo; to return to Pumpkin.&nbsp; Ensure&nbsp;&nbsp; the Pumpkin Server is running by verifying in the bottom right corner of the Pumpkin window.  3)&nbsp;&nbsp;&nbsp; Power on your Polycom Telephone System. 4)&nbsp;&nbsp;&nbsp; As the unit initializes, you will be given the option to Start, go to Setup, or Exit.&nbsp; Chose the setup option on the telephone. a.&nbsp;&nbsp;&nbsp; When prompted enter your administrator password.&nbsp; The default Polycom Password is 456. 5)&nbsp;&nbsp;&nbsp; You will then be taken to a setup menu.&nbsp; Scroll down and select the SERVER MENU option.&nbsp;  6)&nbsp;&nbsp;&nbsp; Once inside the SERVER MENU, select the SERVER TYPE option and change it to TRIVIAL FTP.&nbsp; Save your choice and scroll down to the SERVER ADDERESS option.&nbsp; Enter the IP of the Computer running your Pumpkin Server.&nbsp;  7)&nbsp;&nbsp;&nbsp; To save your changes, continue to select exit until you are prompted to save or discard your changes.&nbsp; Select the SAVE AND REBOOT option. 8)&nbsp;&nbsp;&nbsp; Your Polycom Telephone will reboot and begin to update. This will take as long as fifteen to thirty minutes.&nbsp;  9)&nbsp;&nbsp;&nbsp; Congratulations, you have completed the loading of firmware on your Polycom Telephone system! ]]></contentstext>
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<subject><![CDATA[How to Install Firmware on a Polycom Phone]]></subject>
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<contents><![CDATA[<p>If you ever find yourself locked out of a Yealink phone, and do not know the admin password, you can quickly reset the phone to Factory Default Settings.</p>
<p>Simply press and hold the 'OK' key on the face of the phone for 10 seconds.&nbsp; This will prompt you to accept or decline a factory reset.&nbsp; If you accept, the unit iwll restore itself to factory default settings - without the need the admin password.</p>
<p>The entire process takes roughly 30 seconds.&nbsp;</p>]]></contents>
<contentstext><![CDATA[If you ever find yourself locked out of a Yealink phone, and do not know the admin password, you can quickly reset the phone to Factory Default Settings.
Simply press and hold the 'OK' key on the face of the phone for 10 seconds.&nbsp; This will prompt you to accept or decline a factory reset.&nbsp; If you accept, the unit iwll restore itself to factory default settings - without the need the admin password.
The entire process takes roughly 30 seconds.&nbsp;]]></contentstext>
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<subject><![CDATA[How to Factory Reset a Yealink Phone without the Admin Password]]></subject>
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<contents><![CDATA[Attached is a guide, as well as the tool, to edit the logo on a Yealink IP Telephone]]></contents>
<contentstext><![CDATA[Attached is a guide, as well as the tool, to edit the logo on a Yealink IP Telephone]]></contentstext>
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<contents><![CDATA[This allows a single user to have multiple devices with the same phone  number. It also allows a single device to be used by multiple users in a  hot desk style environment, where a user logs into the device resulting  in it becoming their phone.<br /> <br /> From the CLI navigate to Asterisk directory.<br /> <br /> With your prefered editor ( i like joe or nano ) open amportal.conf and find the following:<br /> <br /> # AMPEXTENSIONS: the type of view for extensions admin<br /> <br /> # If set to 'deviceanduser' Devices and Users will be administered seperately, and Users will be able to "login" to devices.<br /> <br /> # If set to 'extensions' Devices and Users will me administered in a single screen.<br /> <br /> AMPEXTENSIONS=extensions<br /> <br /> Change the value from extensions to deviceanduser like so:<br /> <br /> AMPEXTENSIONS=deviceanduser<br /> <br /> Save and exit.<br /> <br /> Now we need to restart asterisk and amportal. From CLI i use the following command<br /> <br /> amportal restart<br /> <br /> Login to the webui, and create a user. Complete the form to your needs.  Be sure that you supply a password in the User Password field. Now that  we have a user, submit the changes.<br /> <br /> Now we need to create a device. Under Device Options: Provide a  password. Under Device Info: Supply a device ID and Description. For  device type, select Adhoc. For default user select: none.<br /> <br /> Now that we have a device, submit the changes and reload. The user and  device has been created. Now, onto the queue. For creation of the Queue,  please refer here<br /> <br /> <a href="http://freepbx.org/support/documenta...ntation/queues%3Chttp://freepbx.org/support/documentation/module-documentation/queues%3E" target="_blank">http://freepbx.org/support/documenta...ntation/queues&lt;http://freepbx.org/support/documentation/module-documentation/queues&gt;</a><br /> <br /> Place the ID of the USER into the static agents field. Now that we have a queue, submit the changes and reload.<br /> <br /> On the Device, place a call to *11, when prompted, enter your USER  Extension, next it will ask for a password, enter the USER password.  Your User is now an Agent in the Queue.To logout, simply dial *12<br /> <br /> So, with this configuration you should have a device that a user can login or out of at will.]]></contents>
<contentstext><![CDATA[This allows a single user to have multiple devices with the same phone  number. It also allows a single device to be used by multiple users in a  hot desk style environment, where a user logs into the device resulting  in it becoming their phone.    From the CLI navigate to Asterisk directory.    With your prefered editor ( i like joe or nano ) open amportal.conf and find the following:    # AMPEXTENSIONS: the type of view for extensions admin    # If set to 'deviceanduser' Devices and Users will be administered seperately, and Users will be able to "login" to devices.    # If set to 'extensions' Devices and Users will me administered in a single screen.    AMPEXTENSIONS=extensions    Change the value from extensions to deviceanduser like so:    AMPEXTENSIONS=deviceanduser    Save and exit.    Now we need to restart asterisk and amportal. From CLI i use the following command    amportal restart    Login to the webui, and create a user. Complete the form to your needs.  Be sure that you supply a password in the User Password field. Now that  we have a user, submit the changes.    Now we need to create a device. Under Device Options: Provide a  password. Under Device Info: Supply a device ID and Description. For  device type, select Adhoc. For default user select: none.    Now that we have a device, submit the changes and reload. The user and  device has been created. Now, onto the queue. For creation of the Queue,  please refer here    http://freepbx.org/support/documenta...ntation/queues&lt;http://freepbx.org/support/documentation/module-documentation/queues&gt;    Place the ID of the USER into the static agents field. Now that we have a queue, submit the changes and reload.    On the Device, place a call to *11, when prompted, enter your USER  Extension, next it will ask for a password, enter the USER password.  Your User is now an Agent in the Queue.To logout, simply dial *12    So, with this configuration you should have a device that a user can login or out of at will.]]></contentstext>
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<contents><![CDATA[Configuration Guide for 3CX and Bell Commander.]]></contents>
<contentstext><![CDATA[Configuration Guide for 3CX and Bell Commander.]]></contentstext>
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<contents><![CDATA[Trixbox / Asterisk Bell Commander Configuration Guide]]></contents>
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<contents><![CDATA[Cisco Call Manager Express Bell Commander Configuration Guide]]></contents>
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<contents><![CDATA[<p><a href="http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip">Polycom UCS 3.3.0 Split Firmware Download</a></p>
<p><a href="http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip">http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip</a></p>]]></contents>
<contentstext><![CDATA[

	Polycom UCS 3.3.0 Split Firmware Download [1] 

	http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
[2]

Links:
------
[1]
http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
[2]
http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
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<contents><![CDATA[<p><a href="http://www.888voip.com/downloads/polycom/spip_ssip_BootROM_4_3_0_release_sig.zip">Polycom SPIP SSIP BootROM 4.3.0 Download</a></p>
<p><a href="http://www.888voip.com/downloads/polycom/spip_ssip_BootROM_4_3_0_release_sig.zip">http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip</a></p>]]></contents>
<contentstext><![CDATA[

	Polycom SPIP SSIP BootROM 4.3.0 Download [1] 

	http://www.888voip.com/downloads/polycom/UCS_3_3_0_release_sig_split.zip
[2]

Links:
------
[1]
http://www.888voip.com/downloads/polycom/spip_ssip_BootROM_4_3_0_release_sig.zip
[2]
http://www.888voip.com/downloads/polycom/spip_ssip_BootROM_4_3_0_release_sig.zip
]]></contentstext>
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<contents><![CDATA[<p>snom 300 8.4.32 SIP Release Candidate 8 Firmware.</p>
<p>snom300-8.4.32-SIP-RC8-f.bin</p>]]></contents>
<contentstext><![CDATA[

	snom 300 8.4.32 SIP Release Candidate 8 Firmware. 

	snom300-8.4.32-SIP-RC8-f.bin]]></contentstext>
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<contents><![CDATA[snom 320 8.4.32 SIP Release Candidate 8 Firmware]]></contents>
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<contents><![CDATA[<div>snom 360 8.4.32 SIP Release Candidate 8 Firmware</div>]]></contents>
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<contents><![CDATA[<div>snom 370 8.4.32 SIP Release Candidate 8 Firmware</div>]]></contents>
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<div>snom MP 8.4.32 SIP Release Candidate 8 Firmware</div>
</div>]]></contents>
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<contents><![CDATA[<div>
<div>
<div>snom PA1 8.4.32 SIP Release Candidate 8 Firmware</div>
</div>
</div>]]></contents>
<contentstext><![CDATA[  snom PA1 8.4.32 SIP Release Candidate 8 Firmware  ]]></contentstext>
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<contents><![CDATA[<div><a href="http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin">http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin</a></div>
<div><a href="http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin"><br /></a></div>
<div><a href="http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin">snom 820 8.4.32 SIP Release Candidate 8 Firmware</a></div>
<div>
<div>
<div></div>
</div>
</div>]]></contents>
<contentstext><![CDATA[http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin [1]

 snom 820 8.4.32 SIP Release Candidate 8 Firmware [2]     

Links:
------
[1] http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin
[2] http://www.888voip.com/downloads/snom/snom820-8.4.32-SIP-RC8-r.bin
]]></contentstext>
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<contents><![CDATA[<p><a href="http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin">http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin</a></p>
<p><a href="http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin">snom 870 8.4.32 SIP Release Candidate 8 Firmware</a></p>]]></contents>
<contentstext><![CDATA[

	http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin
[1] 

	snom 870 8.4.32 SIP Release Candidate 8 Firmware [2]

Links:
------
[1] http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin
[2] http://www.888voip.com/downloads/snom/snom870-8.4.32-SIP-RC8-r.bin
]]></contentstext>
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<contents><![CDATA[This is an archive of example documents to auto provision Yealink VoIP SIP Phones.]]></contents>
<contentstext><![CDATA[This is an archive of example documents to auto provision Yealink VoIP
SIP Phones.]]></contentstext>
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<subject><![CDATA[Yealink Auto Provision Startup Example]]></subject>
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<contents><![CDATA[<p style="text-align: left;"><strong>Overview:</strong> The <a href="/products/yealink-ehs36.html">Yealink EHS36</a> is an Electronic Hook Switch adapter, which allows you to control the &ldquo;on/off hook&rdquo; status of your <a href="/products/yealink-phones/">Yealink IP Phone</a>, from your headset. With this device, you simply need to press your <a href="/products/voip-headsets/">headset</a> function button to answer or hang-up calls. This allows you to easily answer or hang-up calls when you are not within arms reach of your phone.</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/rsz_1yealink_ehs36_zoom.jpg" alt="EHS36" width="248" height="186" /></p>
<p><strong>Set-up Guide:</strong> The set-up of this device with your supported Plantronics headset and Yealink phone is fairly simple. For a list of supported Headsets, please reference this link: <a href="/downloads/kb/Tested_headset_list_compatible_with_yealink_phone.pdf">Supported Headsets</a></p>
<p>&nbsp;To begin, you will need the following cords: (these cords will come with your headset and EHS36)</p>
<ul>
<li>1 3.5mm cord</li>
<li>1 RJ11 cord</li>
<li>1 RJ12 cord</li>
</ul>
<p>&nbsp;</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/yealink-ehs36-plantronics-before-599.png" alt="" width="599" height="505" /></p>
<p style="text-align: center;"><em>Everything you will need</em></p>
<p style="text-align: center;">&nbsp;</p>
<p style="text-align: left;"><strong>Step 1: </strong>Connect the 3.5mm cord to the EHS36, and the Plantronics Base. This cord will send the pickup/hang-up signal to the EHS36.</p>
<p style="text-align: left;">&nbsp;</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/rsz_20141114_100111.jpg" alt="" width="600" height="338" /></p>
<p>&nbsp;</p>
<p><strong>Step 2: </strong>Connect the RJ12 cord to the EHS36, and into the EXT port on the phone. This cord will send the pickup/hang-up signal to the phone.</p>
<p>&nbsp;</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/rsz_20141114_100216.jpg" alt="" width="600" height="338" /></p>
<p>&nbsp;</p>
<p><strong>Step 3: </strong>Connect the RJ11 cord to the phones headset port, and to the Plantronics base. This cord will send audio between your headset, and phone.</p>
<p>&nbsp;</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/rsz_20141114_100340.jpg" alt="" width="600" height="338" /></p>
<p style="text-align: center;">&nbsp;</p>
<p>&nbsp;Congratulations! You have completed the set-up of the EHS36, with your Plantronics headset and Yealink device.</p>
<p>&nbsp;</p>
<p style="text-align: center;"><img src="/wp-content/uploads/2015/01/Yealink-ehs36-plantronics-after-600.png" alt="" width="600" height="409" /></p>
<p style="text-align: center;">&nbsp;<em>Completed Setup</em></p>
<p style="text-align: center;">&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p><strong>&nbsp;</strong></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: The Yealink EHS36
[http://www.888voip.com/products/yealink-ehs36.html] is an Electronic
Hook Switch adapter, which allows you to control the "on/off hook"
status of your Yealink IP Phone
[http://www.888voip.com/products/yealink-phones/], from your headset.
With this device, you simply need to press your headset
[http://www.888voip.com/products/voip-headsets/] function button to
answer or hang-up calls. This allows you to easily answer or hang-up
calls when you are not within arms reach of your phone. 

SET-UP GUIDE: The set-up of this device with your supported
Plantronics headset and Yealink phone is fairly simple. For a list of
supported Headsets, please reference this link: Supported Headsets
[http://www.888voip.com/downloads/kb/Tested_headset_list_compatible_with_yealink_phone.pdf]


 To begin, you will need the following cords: (these cords will come
with your headset and EHS36) 

 	* 1 3.5mm cord
 	* 1 RJ11 cord
 	* 1 RJ12 cord

_Everything you will need_ 

STEP 1: Connect the 3.5mm cord to the EHS36, and the Plantronics Base.
This cord will send the pickup/hang-up signal to the EHS36. 

STEP 2: Connect the RJ12 cord to the EHS36, and into the EXT port on
the phone. This cord will send the pickup/hang-up signal to the phone.


STEP 3: Connect the RJ11 cord to the phones headset port, and to the
Plantronics base. This cord will send audio between your headset, and
phone. 

 Congratulations! You have completed the set-up of the EHS36, with
your Plantronics headset and Yealink device. 

 _Completed Setup_ 

]]></contentstext>
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<subject><![CDATA[Installation Guide: Yealink EHS36 with Plantronics Headset and Yealink IP Phone]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong> A VoIP provider is a company who provides Voice over Internet Protocol services, more commonly know as phone service delivered over an internet connection. VoIP providers are an increasingly popular method of delivering dial tone to 3CX Phone System.</p>
<p>VoIP providers who have completed 3CX's certification tests to prove their service works as expected with 3CX Phone System can be found in the drop-down menu of supported VoIP providers within the 3CX Installation Wizard and 3CX Management Console. These supported VoIP providers, organized by region, are incredibly easy to set-up and configure with most of the settings already built in. If your provider is not on the list, you can manually enter the required information as a generic VoIP provider.</p>
<p>To check your bandwidth's VoIP call capabilites or determine your bandwidth requirements based on your projected voice traffic, please reference this handy <a href="http://www.3cx.com/blog/docs/bandwidth-utilised-for-voip/">VoIP bandwidth usage guide</a>.</p>
<p>The types of VoIP connections made from a Phone System to a VoIP provider typically fall into one of two categories:</p>
<p><em>SIP Trunk:</em> These types of connections generally do not require registration from the PBX. The PBX&rsquo;s IP needs to be configured with provider, where they will route incoming calls to your PBX.</p>
<p><em>SIP Provider:</em> These types of connections require the PBX to register to them, with an authentication ID and Password.</p>
<p><strong>Configuration:</strong> In this configuration, we are going to use a supported VoIP provider in the US. Please make sure all Firewall configurations are complete, before starting: <a href="http://www.3cx.com/docs/firewall-router-configuration-voip/">Firewall Configuration</a>. If you are using an unsupported provider, please have all of your account &amp; registration information handy. Please reference this link for a list of supported providers:&nbsp;<a href="http://www.3cx.com/partners/voip-providers/">Supported SIP Providers</a></p>
<p><strong>Step 1:</strong> Navigate to the VoIP Providers tab, and click Add Provider</p>
<p><img src="/wp-content/uploads/2015/01/rsz_1.jpg" alt="" width="600" height="432" /></p>
<p><strong>Step 2: </strong>Choose a unique name for your provider, since 3CX is fully field upgradable and it is likely that you will be expanding this phone system in the future. If you are using a supported VoIP provider, select your country and select your provider. If you are manually entering an unsupported VoIP provider, select Generic and either Generic SIP Trunk or Generic VoIP provider depending if your VoIP provider requires a registration from the PBX.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_21.jpg" alt="" width="600" height="377" /></p>
<p><strong>Step 3:</strong> If you are using a supported provider, 3CX will automatically fill in these fields. If you are using an unsupported provider, please reference your account &amp; registration information.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_31.jpg" alt="" width="600" height="398" /></p>
<p><strong>Step 4</strong>: Enter the accounts main phone number, Authenticating ID and authentication password. For SIP Trunks, you will not need an authentication ID, or password. Please enter the number of simultaneous calls you can utilize through this connection, determined by your VoIP provider. Ideally, this should be less than or equal to your 3CX license size.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_4.jpg" alt="" width="600" height="387" /></p>
<p><strong>Step 5: </strong>Configure the inbound routing logic for this VoIP provider. This is for the main account number, which we will send to extension 100 both during and out of office hours. Each DID (Direct Inward Dial) number you obtain from your VoIP provider can be configured with its own inbound routing logic as well. Each radio button will allow you to route your call to any 3CX extension within its extension group. You may set a different call from for after business hours, or keep the call behavior the same. Simply uncheck &ldquo;Same as out of office hours&rdquo; to route calls differently outside of office hours.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_5.jpg" alt="" width="600" height="358" /></p>
<p><strong>Step 6: </strong>Next, we will configure the outbound rules to tell 3CX Phone System when to route outbound calls through our new VoIP provider. Below is a good starting point, allowing all extensions to dial a 7, 10 or 11 digit number and be routed through the Provider we just created, Route 1. For best call control, you will need to set up multiple rules. You can create a basic rule now, or skip this step to design the company&rsquo;s entire required call flow later on.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_6.jpg" alt="" width="600" height="311" /></p>
<p>Congratulations, you have configured a VoIP Provider with 3CX Phone System!</p>]]></contents>
<contentstext><![CDATA[OVERVIEW: A VoIP provider is a company who provides Voice over
Internet Protocol services, more commonly know as phone service
delivered over an internet connection. VoIP providers are an
increasingly popular method of delivering dial tone to 3CX Phone
System. 

VoIP providers who have completed 3CX's certification tests to prove
their service works as expected with 3CX Phone System can be found in
the drop-down menu of supported VoIP providers within the 3CX
Installation Wizard and 3CX Management Console. These supported VoIP
providers, organized by region, are incredibly easy to set-up and
configure with most of the settings already built in. If your provider
is not on the list, you can manually enter the required information as
a generic VoIP provider. 

To check your bandwidth's VoIP call capabilites or determine your
bandwidth requirements based on your projected voice traffic, please
reference this handy VoIP bandwidth usage guide
[http://www.3cx.com/blog/docs/bandwidth-utilised-for-voip/]. 

The types of VoIP connections made from a Phone System to a VoIP
provider typically fall into one of two categories: 

_SIP Trunk:_ These types of connections generally do not require
registration from the PBX. The PBX's IP needs to be configured with
provider, where they will route incoming calls to your PBX. 

_SIP Provider:_ These types of connections require the PBX to register
to them, with an authentication ID and Password. 

CONFIGURATION: In this configuration, we are going to use a supported
VoIP provider in the US. Please make sure all Firewall configurations
are complete, before starting: Firewall Configuration
[http://www.3cx.com/docs/firewall-router-configuration-voip/]. If you
are using an unsupported provider, please have all of your account &
registration information handy. Please reference this link for a list
of supported providers: Supported SIP Providers
[http://www.3cx.com/partners/voip-providers/] 

STEP 1: Navigate to the VoIP Providers tab, and click Add Provider 

STEP 2: Choose a unique name for your provider, since 3CX is fully
field upgradable and it is likely that you will be expanding this
phone system in the future. If you are using a supported VoIP
provider, select your country and select your provider. If you are
manually entering an unsupported VoIP provider, select Generic and
either Generic SIP Trunk or Generic VoIP provider depending if your
VoIP provider requires a registration from the PBX. 

STEP 3: If you are using a supported provider, 3CX will automatically
fill in these fields. If you are using an unsupported provider, please
reference your account & registration information. 

STEP 4: Enter the accounts main phone number, Authenticating ID and
authentication password. For SIP Trunks, you will not need an
authentication ID, or password. Please enter the number of
simultaneous calls you can utilize through this connection, determined
by your VoIP provider. Ideally, this should be less than or equal to
your 3CX license size. 

STEP 5: Configure the inbound routing logic for this VoIP provider.
This is for the main account number, which we will send to extension
100 both during and out of office hours. Each DID (Direct Inward Dial)
number you obtain from your VoIP provider can be configured with its
own inbound routing logic as well. Each radio button will allow you to
route your call to any 3CX extension within its extension group. You
may set a different call from for after business hours, or keep the
call behavior the same. Simply uncheck "Same as out of office hours"
to route calls differently outside of office hours. 

STEP 6: Next, we will configure the outbound rules to tell 3CX Phone
System when to route outbound calls through our new VoIP provider.
Below is a good starting point, allowing all extensions to dial a 7,
10 or 11 digit number and be routed through the Provider we just
created, Route 1. For best call control, you will need to set up
multiple rules. You can create a basic rule now, or skip this step to
design the company's entire required call flow later on. 

Congratulations, you have configured a VoIP Provider with 3CX Phone
System!]]></contentstext>
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<author />
<email />
<subject><![CDATA[Configuration Guide: Adding a VoIP Provider with 3CX]]></subject>
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<contents><![CDATA[<p><a href="http://www.888voip.com/wp-content/uploads/2014/07/3CX-and-Yealink.jpg"><img class="alignright size-small wp-image-13155" title="3CX and Yealink" src="http://www.888voip.com/wp-content/uploads/2014/07/3CX-and-Yealink-200x128.jpg" alt="3CX and Yealink" width="150" height="96" /></a></p>
<p><strong>Overview:&nbsp;</strong>In this configuration guide, we will adjust the outbound gain of a Yealink T4X Phone registered to <a href="/products/3cx-phone-system/">3CX Phone System</a>. We will accomplish this by customizing our T4 Provisioning Template. By changing our provisioning template, we can easily apply our custom changes to all of the Yealink T4X Phones registered to our system.</p>
<p>The same syntax changes can be applied to the configuration templates of Yealink T4X Phones registered to other phone systems, although the steps will vary from what is show here.</p>
<p>There are a few symptoms your clients may report that indicate you need to adjust the outbound gain on your Yealink T4X Phones:</p>
<ol>
<li>
<p><strong>The remote party cannot hear your client well.</strong></p>
<p>This scenario indicates that the outbound volume on your Yealink T4X Phone is set too low, and you need to increase the outbound gain.</p>
</li>
<li>
<p><strong>Your client sounds abrasive and distorted to the remote party.</strong></p>
<p>This scenario gives the illusion that your phone has poor audio quality. However, this simply means that your outbound audio gain is set too high. Lowering the outbound gain is the key here.</p>
</li>
<li>
<p><strong>Your client reports echo on their calls.</strong></p>
<p>This scenario occurs when your client's audio is so loud that the remote party creates an acoustic echo during the call. This acoustic echo is caused by your client's audio projecting from the remote party's handset speaker right back to the remote party's handset microphone, occasionally combined with the echo of the room the remote party is in. Simply lower your client's outbound gain to remove this acoustic echo.&nbsp;</p>
</li>
</ol>
<p><strong>Configuration Guide:&nbsp;</strong>Adjusting the outbound gain of your Yealink T4X Phone can only be done by altering the phone's configuration file. If you are using&nbsp;<a href="/products/3cx-phone-system/">3CX</a>, we recommend watching this quick tutorial on&nbsp;<a href="http://www.3cx.com/3CXAcademy/Videos/Advanced/Provisioning-Templates/" target="_blank">Provisioning&nbsp;Template modifications</a>.</p>
<p>You will need to add the following parameters to your configuration file:</p>
<p class="codeexample">#Configure the sending volume of Speaker, Handset and Headset. It ranges from 1 to 53, the default values are 25, 35, 29.<br /> #Require reboot;<br /> voice.handfree_send =&nbsp;25<br /> voice.handset_send = 35<br /> voice.headset_send = 29</p>
<p>&nbsp;We have also included the finished configuration files in .txt format here, so you can check your work and see how the finished configuration files will look.</p>
<ol>
<li><a href="http://www.888voip.com/wp-content/uploads/2015/06/yealinkt4xcustom_volume.txt">yealinkt4xcustom_volume</a></li>
<li><a href="http://www.888voip.com/wp-content/uploads/2015/06/yealinkt46t48custom_volume.txt">yealinkt46t48custom_volume</a></li>
</ol>
<p><strong>Step 1:&nbsp;</strong>First, we need to locate the provisioning templates in order to customize them. Please ensure you can see your hidden files and folders, then navigate to "C:/ProgramData/3CX/Data/Http/Templates/phones". Next, copy the default templates "yealinkT4x.ph.xml" and "yealinkT46T48.ph.xml" and rename them as shown in the screenshot below:</p>
<p><img src="https://www.888voip.com/wp-content/uploads/2015/01/rsz__1.jpg" alt="" width="600" height="455" /></p>
<p>We edit a copy of the standard provisioning template instead of editing the original so our custom changes are not overwritten then next time a service pack with updated templates is released.&nbsp;</p>
<p><strong>Step 2:</strong>&nbsp;Now that we have our custom template, we can continue towards adds our custom parameters. Now is the time to open your custom template in your favorite .XML editor. Here at 888VoIP, we like to use&nbsp;<a href="http://notepad-plus-plus.org/">Notepad++</a>.</p>
<p>The first thing to edit in our custom template is the contents of the &lt;models&gt; tag. This dictates what is shown in the dropdown menu of the provisioning tab in the 3CX Management Console, and it is important to change so you will know which template to select when provisioning your phones. The &lt;models&gt; tag is located at the beginning of the template:</p>
<p><img src="https://www.888voip.com/wp-content/uploads/2015/01/rsz__4.jpg" alt="" width="600" height="294" /></p>
<p><strong>Step 3:</strong>&nbsp;Next we need to add the outbound gain configuration parameters below:</p>
<p class="codeexample">#Configure the sending volume of Speaker, Handset and Headset. It ranges from 1 to 53, the default values are 25, 35, 29.<br /> #Require reboot;<br /> voice.handfree_send =&nbsp;25<br /> voice.handset_send = 35<br /> voice.headset_send = 29</p>
<p>A common practice for adding custom parameters is to place them at the end of the configuration file, so you can easily locate them if you need to tweak their values or add your custom changes to a new provisioning template that is released in the future.</p>
<p><img src="http://www.888voip.com/wp-content/uploads/2015/01/rsz_3.jpg" alt="" width="600" height="106" /></p>
<p>&nbsp;</p>
<p>Now that you have added the outbound gain parameters to your configuration file, go ahead and practice turning them up and down to familiarize yourself with how the settings affect the audio of your Yealink T4X Phone.</p>
<p>Congratulations, you have&nbsp;successfully adjusted the outbound gain of your Yealink T4X Phone with a custom configuration file! If you have been following along with 3CX Phone System, you will now see your custom template when Auto-Provisioning.</p>
<p><img src="https://www.888voip.com/wp-content/uploads/2015/01/rsz_2.jpg" alt="" width="600" height="411" /></p>]]></contents>
<contentstext><![CDATA[ [http://www.888voip.com/wp-content/uploads/2014/07/3CX-and-Yealink.jpg]


OVERVIEW:In this configuration guide, we will adjust the outbound gain
of a Yealink T4X Phone registered to 3CX Phone System
[http://www.888voip.com/products/3cx-phone-system/]. We will
accomplish this by customizing our T4 Provisioning Template. By
changing our provisioning template, we can easily apply our custom
changes to all of the Yealink T4X Phones registered to our system. 

The same syntax changes can be applied to the configuration templates
of Yealink T4X Phones registered to other phone systems, although the
steps will vary from what is show here. 

There are a few symptoms your clients may report that indicate you
need to adjust the outbound gain on your Yealink T4X Phones: 

	* 

THE REMOTE PARTY CANNOT HEAR YOUR CLIENT WELL. 

This scenario indicates that the outbound volume on your Yealink T4X
Phone is set too low, and you need to increase the outbound gain. 
	* 

YOUR CLIENT SOUNDS ABRASIVE AND DISTORTED TO THE REMOTE PARTY. 

This scenario gives the illusion that your phone has poor audio
quality. However, this simply means that your outbound audio gain is
set too high. Lowering the outbound gain is the key here. 
	* 

YOUR CLIENT REPORTS ECHO ON THEIR CALLS. 

This scenario occurs when your client's audio is so loud that the
remote party creates an acoustic echo during the call. This acoustic
echo is caused by your client's audio projecting from the remote
party's handset speaker right back to the remote party's handset
microphone, occasionally combined with the echo of the room the remote
party is in. Simply lower your client's outbound gain to remove this
acoustic echo. 

CONFIGURATION GUIDE:Adjusting the outbound gain of your Yealink T4X
Phone can only be done by altering the phone's configuration file. If
you are using 3CX [http://www.888voip.com/products/3cx-phone-system/],
we recommend watching this quick tutorial on Provisioning Template
modifications
[http://www.3cx.com/3CXAcademy/Videos/Advanced/Provisioning-Templates/].


You will need to add the following parameters to your configuration
file: 

#Configure the sending volume of Speaker, Handset and Headset. It
ranges from 1 to 53, the default values are 25, 35, 29.
 #Require reboot;
 voice.handfree_send = 25
 voice.handset_send = 35
 voice.headset_send = 29 

 We have also included the finished configuration files in .txt format
here, so you can check your work and see how the finished
configuration files will look. 

 	* yealinkt4xcustom_volume
[http://www.888voip.com/wp-content/uploads/2015/06/yealinkt4xcustom_volume.txt]
 	* yealinkt46t48custom_volume
[http://www.888voip.com/wp-content/uploads/2015/06/yealinkt46t48custom_volume.txt]

STEP 1:First, we need to locate the provisioning templates in order to
customize them. Please ensure you can see your hidden files and
folders, then navigate to
"C:/ProgramData/3CX/Data/Http/Templates/phones". Next, copy the
default templates "yealinkT4x.ph.xml" and "yealinkT46T48.ph.xml" and
rename them as shown in the screenshot below: 

We edit a copy of the standard provisioning template instead of
editing the original so our custom changes are not overwritten then
next time a service pack with updated templates is released. 

STEP 2: Now that we have our custom template, we can continue towards
adds our custom parameters. Now is the time to open your custom
template in your favorite .XML editor. Here at 888VoIP, we like to use
Notepad++ [http://notepad-plus-plus.org/]. 

The first thing to edit in our custom template is the contents of the
<models> tag. This dictates what is shown in the dropdown menu of the
provisioning tab in the 3CX Management Console, and it is important to
change so you will know which template to select when provisioning
your phones. The <models> tag is located at the beginning of the
template: 

STEP 3: Next we need to add the outbound gain configuration parameters
below: 

#Configure the sending volume of Speaker, Handset and Headset. It
ranges from 1 to 53, the default values are 25, 35, 29.
 #Require reboot;
 voice.handfree_send = 25
 voice.handset_send = 35
 voice.headset_send = 29 

A common practice for adding custom parameters is to place them at the
end of the configuration file, so you can easily locate them if you
need to tweak their values or add your custom changes to a new
provisioning template that is released in the future. 

Now that you have added the outbound gain parameters to your
configuration file, go ahead and practice turning them up and down to
familiarize yourself with how the settings affect the audio of your
Yealink T4X Phone. 

Congratulations, you have successfully adjusted the outbound gain of
your Yealink T4X Phone with a custom configuration file! If you have
been following along with 3CX Phone System, you will now see your
custom template when Auto-Provisioning. 

]]></contentstext>
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<subject><![CDATA[Configuration Guide: Adjusting your T4X's Outbound Volume ]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong>&nbsp;In this guide, we will show you how to adjust the contrast of your expansion module. Sometimes the expansion module may appear to be defective, because it is very light or dark, compared to the phone. This is because the expansion modules contrast, adjust separately from the phone.&nbsp;</p>
<p>&nbsp;</p>
<p><strong>Configuration Guide:</strong><span class="apple-converted-space">&nbsp;</span>To use the expansion module with your device, please make sure it is on the proper<span class="apple-converted-space">&nbsp;</span><a href="http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip_expansion_module.html" target="_blank">firmware</a>. Adjusting the contrast is done in six simple steps.</p>
<p>&nbsp;</p>
<p><strong>Step 1:&nbsp;</strong>After powering on your device, Please navigate to the phones Menu options.</p>
<p><strong><img src="/wp-content/uploads/2015/01/rsz_polycom_expasion_step1.jpg" alt="" width="600" height="338" /></strong></p>
<p>&nbsp;</p>
<p><strong>Step 2:&nbsp;</strong>Then use the down arrow to navigate to "Settings" and press the check mark button, for ok.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_polycom_expasion_step2.jpg" alt="" width="600" height="338" /></p>
<p>&nbsp;</p>
<p><strong>Step 3:&nbsp;</strong>Press the check mark button again, while "Basic" is highlighted.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_polycom_expasion_step3.jpg" alt="" width="600" height="338" /></p>
<p>&nbsp;</p>
<p><strong>Step 4:&nbsp;</strong>Press the down arrow to select "Contrast", and press the check mark button.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_polycom_expasion_step4.jpg" alt="" width="600" height="338" /></p>
<p>&nbsp;</p>
<p><strong>Step 5:&nbsp;</strong>Here you will see "UP" and "Down" on the expansion module. Please use the buttons directly to the right of these words. These will adjust the&nbsp;expansion modules contrast, separately from the phone.</p>
<p><img src="/wp-content/uploads/2015/01/rsz_polycom_expasion_step5.jpg" alt="" width="600" height="338" /></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: In this guide, we will show you how to adjust the contrast
of your expansion module. Sometimes the expansion module may appear to
be defective, because it is very light or dark, compared to the phone.
This is because the expansion modules contrast, adjust separately from
the phone. 

CONFIGURATION GUIDE: To use the expansion module with your device,
please make sure it is on the proper firmware
[http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip_expansion_module.html].
Adjusting the contrast is done in six simple steps. 

STEP 1:After powering on your device, Please navigate to the phones
Menu options. 

STEP 2:Then use the down arrow to navigate to "Settings" and press the
check mark button, for ok. 

STEP 3:Press the check mark button again, while "Basic" is
highlighted. 

STEP 4:Press the down arrow to select "Contrast", and press the check
mark button. 

STEP 5:Here you will see "UP" and "Down" on the expansion module.
Please use the buttons directly to the right of these words. These
will adjust the expansion modules contrast, separately from the phone.


]]></contentstext>
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<subject><![CDATA[Configuration Guide: Adjusting the Contrast on Polycom IP Series Expansion Modules]]></subject>
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<contents><![CDATA[<p>If you're having trouble activating&nbsp;your 3CX license key, you may be seeing an error similar to this:</p>
<p><strong>"</strong><strong>License Key Activation Failed License:18 System:12029"<br /></strong></p>
<p>If you see an error like this, make sure to check which email addres you're trying to activate your key with. You cannot activate your key with the same email address used in your 3CX Partner Portal.</p>
<p>You will need to use your customer's email address to activate the license key. Use a different email address, then try activating your key again, you should be able to activate without any errors.</p>
<p>&nbsp;</p>]]></contents>
<contentstext><![CDATA[If you're having trouble activating your 3CX license key, you may be
seeing an error similar to this: 

"License Key Activation Failed License:18 System:12029"

If you see an error like this, make sure to check which email addres
you're trying to activate your key with. You cannot activate your key
with the same email address used in your 3CX Partner Portal. 

You will need to use your customer's email address to activate the
license key. Use a different email address, then try activating your
key again, you should be able to activate without any errors. 

 ]]></contentstext>
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<subject><![CDATA[Why Can't I Activate My 3CX License Key?]]></subject>
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<contents><![CDATA[<p>Having problems registering remote phones, or can't access 3CX Company Phonebook on a remote phone?</p>
<p>Check to make sure your phone was provisioned with the remote 3CX provisioning URL. If you provisioned the phone on the local network, then moved the phone to the remote site, it most likely is using the local provisioning URL.</p>
<p>In this case, re-provision the phone using the remote provisioning URL.</p>]]></contents>
<contentstext><![CDATA[Having problems registering remote phones, or can't access 3CX Company
Phonebook on a remote phone? 

Check to make sure your phone was provisioned with the remote 3CX
provisioning URL. If you provisioned the phone on the local network,
then moved the phone to the remote site, it most likely is using the
local provisioning URL. 

In this case, re-provision the phone using the remote provisioning
URL.]]></contentstext>
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<subject><![CDATA[Why won't my remote phone register?]]></subject>
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<contents><![CDATA[<p>Installation Guide: Yealink EHS36 with Sennheiser Headset and Yealink IP Phone</p>
<p><strong>Overview:</strong>&nbsp;The <a href="https://www.888voip.com/products/yealink-ehs36.html">Yealink EHS36</a> is an Electronic Hook Switch adapter, which allows you to control the &ldquo;on/off hook&rdquo; status of your <a href="https://www.888voip.com/products/yealink-phones/">Yealink IP Phone</a>, from your headset. With this device, you simply need to press your <a href="https://www.888voip.com/products/voip-headsets/">headset</a> function button to answer or hang-up calls. This allows you to easily answer or hang-up calls when you are not within arms reach of your phone.</p>
<p><strong>Set-up Guide: </strong>The set-up of this device with your supported Sennheiser headset and Yealink phone is fairly simple.</p>
<p>&nbsp;</p>
<p>What you will need:</p>
<ol>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Sennheiser Y-Splitter</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Yealink EHS36</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RJ-12 Cord</li>
</ol>
<p>&nbsp;</p>
<ol>
<li>Flip over the Sennheiser headset base and look for 6 dip-switches on the bottom right<ol style="list-style-type: lower-alpha;">
<li>Configure the Fast Link (DHSG, MSH) switches to be toggled to the left side &ndash; These should be the first two dip-switches</li>
<li>All other dip-switches should be toggled to the right hand side</li>
<li><img src="/wp-content/uploads/2016/01/DipSwitch11.jpg" alt="" width="338" /></li>
</ol></li>
</ol>
<p>&nbsp;</p>
<ol start="2">
<li>Plug the DC cable into the Sennheiser headset base<br /><ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/DCPower21.jpg" alt="" width="338" /></li>
</ol></li>
<li>Place the Sennheiser Headset onto the Base</li>
<li>Plug the DC cable into an outlet</li>
<li>Press the button with a Phone image on the Sennheiser Base &ndash; this should be on the right hand side<ol style="list-style-type: lower-alpha;">
<li>Verify that the indicator light on the base appears (See image below)</li>
</ol></li>
<li>Verify that the Sennheiser headset indicator light turns on<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/Headset31.jpg" alt="" width="338" /></li>
</ol></li>
</ol>
<p>&nbsp;</p>
<ol start="7">
<li>Plug the 8-Pin Joint connection from the Y-Cable into the Sennheiser Base<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/SplitterBase41.jpg" alt="" width="338" /></li>
</ol></li>
</ol><ol start="8">
<li>Plug the RJ-45 side of the splitter into the Yealink EHS36<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/SplitterEH51.jpg" alt="" width="338" /></li>
</ol></li>
</ol><ol start="9">
<li>Plug the RJ-11 side of the splitter into the Headset section of the Yealink Desk Phone<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/SplitterPhone61.jpg" alt="" width="338" /></li>
</ol></li>
<li>Plug one end of the double ended RJ-12 cord into the Yealink EHS36&nbsp;<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/RJEH71.jpg" alt="" width="338" /></li>
</ol></li>
<li>Plug the other end of the double ended RJ-12 cord into the Yealink IP Desk Phone EXT Port<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/RJEXT81.jpg" alt="" width="338" /></li>
<li>Test out the headset by Removing the headset from the base and Pressing the Desk Phone button on the Sennheiser Base</li>
</ol></li>
<li>Final Set Up:<ol style="list-style-type: lower-alpha;">
<li><img src="/wp-content/uploads/2016/01/Final92.jpg" alt="" width="338" /></li>
</ol></li>
</ol>]]></contents>
<contentstext><![CDATA[Installation Guide: Yealink EHS36 with Sennheiser Headset and Yealink
IP Phone 

OVERVIEW: The Yealink EHS36
[https://www.888voip.com/products/yealink-ehs36.html] is an Electronic
Hook Switch adapter, which allows you to control the "on/off hook"
status of your Yealink IP Phone
[https://www.888voip.com/products/yealink-phones/], from your headset.
With this device, you simply need to press your headset
[https://www.888voip.com/products/voip-headsets/] function button to
answer or hang-up calls. This allows you to easily answer or hang-up
calls when you are not within arms reach of your phone. 

SET-UP GUIDE: The set-up of this device with your supported Sennheiser
headset and Yealink phone is fairly simple. 

What you will need: 

 	* Sennheiser Y-Splitter
 	* Yealink EHS36
 	* RJ-12 Cord

	* Flip over the Sennheiser headset base and look for 6 dip-switches
on the bottom right

 	* Configure the Fast Link (DHSG, MSH) switches to be toggled to the
left side - These should be the first two dip-switches
 	* All other dip-switches should be toggled to the right hand side
 	* 

	* Plug the DC cable into the Sennheiser headset base

 	* 

 	* Place the Sennheiser Headset onto the Base
 	* Plug the DC cable into an outlet

	* Press the button with a Phone image on the Sennheiser Base - this
should be on the right hand side

 	* Verify that the indicator light on the base appears (See image
below)

	* Verify that the Sennheiser headset indicator light turns on

 	* 

	* Plug the 8-Pin Joint connection from the Y-Cable into the
Sennheiser Base

 	* 

	* Plug the RJ-45 side of the splitter into the Yealink EHS36

 	* 

	* Plug the RJ-11 side of the splitter into the Headset section of the
Yealink Desk Phone

 	* 

	* Plug one end of the double ended RJ-12 cord into the Yealink EHS36 

 	* 

	* Plug the other end of the double ended RJ-12 cord into the Yealink
IP Desk Phone EXT Port

 	* 
 	* Test out the headset by Removing the headset from the base and
Pressing the Desk Phone button on the Sennheiser Base

	* Final Set Up:

 	* 

]]></contentstext>
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<subject><![CDATA[ Installation Guide: Yealink EHS36 with Sennheiser Headset and Yealink IP Phone]]></subject>
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<contents><![CDATA[<p>What is 3CX?:</p>
<p>3CX is a software based PBX (Private Branch Exchange) based on the SIP Standard which enables users to make phone calls internally and externally via PSTN (Public Switched Telephone Network) or VoIP (Voice over Internet Protocol)</p>
<p>Prior to the installation, please verify that your system has the following: Please note that these are the minimum requirements to run the PBX, but additional requirements may be needed for call recording.</p>
<p>If you would like more information on hardware requirements please <a href="http://downloads.3cx.com/downloads/3CX_Phone_System_Operating_System_Specifications.pdf">click here for a full list</a></p>
<ul>
<li>Windows (Standard or Server) &ndash; Must be x64 Bit Architecture<br />&nbsp; &nbsp; -Standard Windows &ndash; Windows 7 or Higher<br />&nbsp; &nbsp; -Server Windows &ndash; Windows Server 2008 R2 or Higher (GUI only)</li>
<li>.NET Framework version 4</li>
<li>Atleast 1GB RAM</li>
<li>Atleast 3.2 GHz Processor</li>
<li>Atleast 30GB Free space on your hard drive.</li>
</ul>
<p>If you are installing 3CX Phone System on a Virtual Machine there are additional requirements:</p>
<ul>
<li>Static MAC Address</li>
<li>Integrated Services</li>
</ul>
<p>For more information on requirements for a Virtual Installation please <a href="http://www.3cx.com/docs/virtual-pbx-installation/">click here for a full list</a></p>
<p>Recommendations:</p>
<ul>
<li>It is recommended to make this a dedicated server for only 3CX as it uses a lot of memory and resources</li>
<li>Internet Explorer, Firefox, or Chrome</li>
<li>3CX Supported VoIP Phones &amp; Gateways</li>
<li>3CX Supported VoIP Providers &amp; SIP Trunks</li>
<li>Static port mappings / Port forwarding</li>
<li>Static public IP address</li>
<li>Installing 3CX Phone System on a machine that also has the following products installed &ndash; Microsoft Exchange Server, SQL Server, SharePoint or DNS Role is not supported.</li>
</ul>
<p>After verifying you have the required software and hardware, you can proceed with the installation of 3CX.</p>
<p>The first screen you will see when launching the executable installation file will be the requirements and recommendations that are listed above. Press Next to both of these screens after confirming you have the proper software and hardware components required.</p>
<p><img src="/wp-content/uploads/2016/01/1.jpg" alt="" width="338" height="264" /><img src="/wp-content/uploads/2016/01/2.jpg" alt="" width="338" height="262" /></p>
<p>The next screen that will be displayed is the End-User License Agreement. After reading through the terms, please select the options on whether or not you accept the terms. If you do not accept the terms within the License Agreement you will not be able to proceed with the installation.</p>
<p><img src="/wp-content/uploads/2016/01/3.jpg" alt="" width="338" height="266" /></p>
<p>You will then need to select the directory in which you would like to install 3CX. The default location for this is going to be on your main hard drive within the Program Files folder. You can change the default location by selecting Browse.</p>
<p><img src="/wp-content/uploads/2016/01/4.jpg" alt="" width="338" height="266" /></p>
<p>After selecting where you would like to install 3CX Software, you will now be prompted on whether you would like to have a standard installation or a virtual installation.</p>
<p><img src="/wp-content/uploads/2016/01/5.jpg" alt="" width="338" height="263" /></p>
<ul>
<li>Standard Installation<br />&nbsp; &nbsp; -This option will be the only available option if you are running a Standard Windows Operating System(Non-Server)</li>
</ul>
<ul>
<li>Virtual PBX Installation<br />&nbsp; &nbsp; -Only available on Windows Server machines<br />&nbsp; &nbsp; -Allows the server to accept up to 25 instances of 3CX on one server</li>
</ul>
<p>If you are installing 3CX on a server behind a NAT, you will have to configure port forwarding to allow traffic to come through to your server and be able to communicate with 3CX properly. If your server is not behind a NAT, you will not need to do additional port forwarding but still may need to verify that your firewall is not blocking any incoming and outgoing traffic. If you are unsure of which option applies to your setup, please contact your network administrators for more information</p>
<p><img src="/wp-content/uploads/2016/01/6.jpg" alt="" width="338" height="265" /></p>
<p>If you select No NAT you will be presented with the next screen only asking for the default network interface. This is a drop down menu which allows you to select the proper IP address if you have more than one Network Interface Card.</p>
<p><img src="/wp-content/uploads/2016/01/7.jpg" alt="" width="338" height="264" /></p>
<p>If you selected the option that you are behind a NAT then you will have to select both the Default Network Card as well as the Static External (public) IP Address.</p>
<p><img src="/wp-content/uploads/2016/01/7_1.jpg" alt="" width="338" height="264" /></p>
<p>After configuring the NAT and IP settings, you will be prompted for an administration email address. You will be notified by email when any updates are available as well as any system notifications such as outages or service failure.</p>
<p>&nbsp;<img src="/wp-content/uploads/2016/01/8.jpg" alt="" width="338" height="265" /></p>
<p>The next step of the 3CX installer will prompt you whether you would like to use:</p>
<ul>
<li>Single FQDN<br />&nbsp; &nbsp; -May require split DNS<br />&nbsp; &nbsp; -May require Dedicated Domain<br />&nbsp; &nbsp; -Only needs a single address</li>
</ul>
<ul>
<li>External/Local FQDN<br />&nbsp; &nbsp; -Needs multiple addresses<br />&nbsp; &nbsp; -May require a domain</li>
</ul>
<ul>
<li>Local IP address<br />&nbsp; &nbsp; -This limits 3CX for internal use only.</li>
</ul>
<p><img src="/wp-content/uploads/2016/01/9.jpg" alt="" width="338" height="265" /></p>
<p>FQDN &ndash; Fully Qualified Domain Name</p>
<ul>
<li>A Fully Qualified Domain Name allows you to have several services running on sub-domains. For example, a mail server may be set up using mail.server.com, pointing to one IP address while a website may be set up under www.server.com, pointing to another IP address. Fully Qualified Domain Names allow different sub domains to route to different IP addresses allowing other services to run on the same hostname.</li>
</ul>
<p>Split DNS (Domain Name Server) &ndash; creates two DNS zones &ndash; one DNS for internal domain name resolution, one DNS for external domain name resolution. In order to set up Split DNS, a dedicated Domain is required.</p>
<ul>
<li>For more information on setting up Split DNS, please look at <a href="http://www.3cx.com/docs/creating-fqdn-split-dns/">3CX&rsquo;s Guide for Creating an FQDN with Split DNS</a></li>
</ul>
<p>The next section, you will need to fill out the information depending on which choice you selected:</p>
<ul>
<li>Single FQDN<br />&nbsp; &nbsp; -Needs only one FQDN</li>
</ul>
<ul>
<li>External and Local FQDN<br />&nbsp; &nbsp; -Needs two FQDNs, one External / Public Domain, and one Internal Domain or Private IP Address</li>
</ul>
<ul>
<li>Local IP<br />&nbsp; &nbsp; -This will allow you to only use your Private IP and Public IP address as the Internal FQDN, making this only available for internal users</li>
</ul>
<p><img src="/wp-content/uploads/2016/01/10.jpg" alt="" width="338" height="266" /></p>
<p>The next step in the installation is setting up outgoing emails. This will be used for any notifications / service failures. You will need to know your Mail Server, your email / reply address, the username that is used to login to email and the password for the account. You then have the option to enable secure messages (SSL/TLS). Please verify with your email administrator that you have SSL/TLS enabled before proceeding.</p>
<p><img src="/wp-content/uploads/2016/01/11.jpg" alt="" width="338" height="263" /></p>
<p>3CX Web Management Console allows you to update settings and features through your internet browser. In order to use this administrative console you need to set up a username and password. This information can be changed through the Management Console</p>
<p><img src="/wp-content/uploads/2016/01/12.jpg" alt="" width="338" height="265" /></p>
<p>In order for the Web Management Console as well as other services to work properly, you need to configure a webserver. The two options that are provided are 3CX Web Server (Abyss Web Server) or Microsoft IIS (Internet Information Service). Abyss will be the only option available if you are installing this on a Standard Windows (Non-Server). IIS is the preferred web server as it is native to the Windows Server Operating System and other services may already be using IIS.</p>
<p><img src="/wp-content/uploads/2016/01/13.jpg" alt="" width="338" height="265" /></p>
<p>Prior to selecting which Web Server you would prefer to use, please verify with your network administrators that nothing is already running on the selected ports or it may cause conflicts and neither application will work. Abyss will allow you to select different port numbers such as 5000(HTTP) and 5001(HTTPS), as seen below. Please note that if you change the ports within Abyss you will not be able to alter this after the installation. IIS will select the web server ports which is 80 (HTTP) and 443 (HTTPS) and will not provide an option to change these ports. If you use IIS please verify that your HTTP (80) and HTTPS (443) ports are not being used by another application or service within the same domain.</p>
<p><img src="/wp-content/uploads/2016/01/14.jpg" alt="" width="338" height="264" /></p>
<p>In order to use the HTTPS port, a SSL Certificate is needed. SSL certificates allow secure communication between web servers and encrypt information to protect data. This encryption is a requirement on 3CX Web Console. If you already have an SSL Certificate you can import it &ndash; but if you do not already have a certificate you can have 3CX installer generate a certificate. Keep in mind, if you select this option it will give security warnings to anyone who will access the webpage.</p>
<p><img src="/wp-content/uploads/2016/01/15.jpg" alt="" width="338" height="261" /></p>
<p>If you selected the option to generate a certificate you will be prompted for you company name as well as a password for your certificate. Please enter this information and press Next to continue with the installation</p>
<p><img src="/wp-content/uploads/2016/01/16.jpg" alt="" width="338" height="265" /></p>
<p>The next screen will show you all of the information for the installation as well as give you the option to change any settings by pressing the Back button. If you are content with your options, you can select Install and the process can take several minutes depending on your hardware specifications</p>
<p><img src="/wp-content/uploads/2016/01/17.jpg" alt="" width="338" height="265" /></p>
<p>After the installation you will have an icon on your desktop. By opening this application, it will open a web address in your default web browser and may take several minutes to load the web page.</p>
<p>The Icon that is on your desktop should look similar to this:</p>
<p><img src="/wp-content/uploads/2016/01/18.jpg" alt="" width="88" height="106" /></p>
<p>After the webpage loads, you will be prompted with a username and a password which was set up during the installation process. Please enter the information and login with your credentials</p>
<p><img src="/wp-content/uploads/2016/01/19.jpg" alt="" width="338" height="251" /></p>
<p>After you are logged into the web management console you will be prompted to set up the number of digits for internal extensions. By default this option is set to 3 digits. Please note that you will not be able to change this setting after the installation is complete. In order to change the extension length a fresh install will need to be performed (uninstalling 3CX and reinstalling). Please note that if you decide to change the extension from a fresh install, you will not be able to restore from a backup because the extension length will be incorrect.</p>
<p><img src="/wp-content/uploads/2016/01/20.jpg" alt="" width="338" height="249" /></p>
<p>You will now be prompted with the time zone and country that your 3CX Phone System is operating from.</p>
<p><img src="/wp-content/uploads/2016/01/21.jpg" alt="" width="338" height="249" /></p>
<p>You will now have to set up an operator extension. This extension is the default destination for inbound calls. Please provide the extension, voicemail extension, their first name, last name, and email address.</p>
<p><img src="/wp-content/uploads/2016/01/22.jpg" alt="" width="338" height="249" /></p>
<p>After setting up your operator extension you will be able to set up options for international calls. You can use the arrows on the left hand side of the region to expand other options and view more specific regions. By checking additional checkboxes, you will allow outbound phone calls to be made to other countries. Any checkbox that is not selected will disallow any external phone calls to be made to that region, but incoming calls from those regions will still be allowed.</p>
<p><img src="/wp-content/uploads/2016/01/23.jpg" alt="" width="338" height="249" /></p>
<p>Your set up is now complete and you should see a screen similar to the image below &ndash; you will also have a quick start guide on the next steps to complete your 3CX Server.</p>
<p>&nbsp;<img src="/wp-content/uploads/2016/01/24.jpg" alt="" width="338" height="151" /></p>]]></contents>
<contentstext><![CDATA[What is 3CX?: 

3CX is a software based PBX (Private Branch Exchange) based on the SIP
Standard which enables users to make phone calls internally and
externally via PSTN (Public Switched Telephone Network) or VoIP (Voice
over Internet Protocol) 

Prior to the installation, please verify that your system has the
following: Please note that these are the minimum requirements to run
the PBX, but additional requirements may be needed for call recording.


If you would like more information on hardware requirements please
click here for a full list
[http://downloads.3cx.com/downloads/3CX_Phone_System_Operating_System_Specifications.pdf]


	* Windows (Standard or Server) - Must be x64 Bit Architecture
 -Standard Windows - Windows 7 or Higher
 -Server Windows - Windows Server 2008 R2 or Higher (GUI only) 	* .NET
Framework version 4
 	* Atleast 1GB RAM
 	* Atleast 3.2 GHz Processor
 	* Atleast 30GB Free space on your hard drive.

If you are installing 3CX Phone System on a Virtual Machine there are
additional requirements: 

 	* Static MAC Address
 	* Integrated Services

For more information on requirements for a Virtual Installation please
click here for a full list
[http://www.3cx.com/docs/virtual-pbx-installation/] 

Recommendations: 

 	* It is recommended to make this a dedicated server for only 3CX as
it uses a lot of memory and resources
 	* Internet Explorer, Firefox, or Chrome
 	* 3CX Supported VoIP Phones & Gateways
 	* 3CX Supported VoIP Providers & SIP Trunks
 	* Static port mappings / Port forwarding
 	* Static public IP address
 	* Installing 3CX Phone System on a machine that also has the
following products installed - Microsoft Exchange Server, SQL Server,
SharePoint or DNS Role is not supported.

After verifying you have the required software and hardware, you can
proceed with the installation of 3CX. 

The first screen you will see when launching the executable
installation file will be the requirements and recommendations that
are listed above. Press Next to both of these screens after confirming
you have the proper software and hardware components required. 

The next screen that will be displayed is the End-User License
Agreement. After reading through the terms, please select the options
on whether or not you accept the terms. If you do not accept the terms
within the License Agreement you will not be able to proceed with the
installation. 

You will then need to select the directory in which you would like to
install 3CX. The default location for this is going to be on your main
hard drive within the Program Files folder. You can change the default
location by selecting Browse. 

After selecting where you would like to install 3CX Software, you will
now be prompted on whether you would like to have a standard
installation or a virtual installation. 

	* Standard Installation
 -This option will be the only available option if you are running a
Standard Windows Operating System(Non-Server) 

	* Virtual PBX Installation
 -Only available on Windows Server machines
 -Allows the server to accept up to 25 instances of 3CX on one server 

If you are installing 3CX on a server behind a NAT, you will have to
configure port forwarding to allow traffic to come through to your
server and be able to communicate with 3CX properly. If your server is
not behind a NAT, you will not need to do additional port forwarding
but still may need to verify that your firewall is not blocking any
incoming and outgoing traffic. If you are unsure of which option
applies to your setup, please contact your network administrators for
more information 

If you select No NAT you will be presented with the next screen only
asking for the default network interface. This is a drop down menu
which allows you to select the proper IP address if you have more than
one Network Interface Card. 

If you selected the option that you are behind a NAT then you will
have to select both the Default Network Card as well as the Static
External (public) IP Address. 

After configuring the NAT and IP settings, you will be prompted for an
administration email address. You will be notified by email when any
updates are available as well as any system notifications such as
outages or service failure. 

The next step of the 3CX installer will prompt you whether you would
like to use: 

	* Single FQDN
 -May require split DNS
 -May require Dedicated Domain
 -Only needs a single address 

	* External/Local FQDN
 -Needs multiple addresses
 -May require a domain 

	* Local IP address
 -This limits 3CX for internal use only. 

FQDN - Fully Qualified Domain Name 

 	* A Fully Qualified Domain Name allows you to have several services
running on sub-domains. For example, a mail server may be set up using
mail.server.com, pointing to one IP address while a website may be set
up under www.server.com, pointing to another IP address. Fully
Qualified Domain Names allow different sub domains to route to
different IP addresses allowing other services to run on the same
hostname.

Split DNS (Domain Name Server) - creates two DNS zones - one DNS for
internal domain name resolution, one DNS for external domain name
resolution. In order to set up Split DNS, a dedicated Domain is
required. 

 	* For more information on setting up Split DNS, please look at 3CX's
Guide for Creating an FQDN with Split DNS
[http://www.3cx.com/docs/creating-fqdn-split-dns/]

The next section, you will need to fill out the information depending
on which choice you selected: 

	* Single FQDN
 -Needs only one FQDN 

	* External and Local FQDN
 -Needs two FQDNs, one External / Public Domain, and one Internal
Domain or Private IP Address 

	* Local IP
 -This will allow you to only use your Private IP and Public IP
address as the Internal FQDN, making this only available for internal
users 

The next step in the installation is setting up outgoing emails. This
will be used for any notifications / service failures. You will need
to know your Mail Server, your email / reply address, the username
that is used to login to email and the password for the account. You
then have the option to enable secure messages (SSL/TLS). Please
verify with your email administrator that you have SSL/TLS enabled
before proceeding. 

3CX Web Management Console allows you to update settings and features
through your internet browser. In order to use this administrative
console you need to set up a username and password. This information
can be changed through the Management Console 

In order for the Web Management Console as well as other services to
work properly, you need to configure a webserver. The two options that
are provided are 3CX Web Server (Abyss Web Server) or Microsoft IIS
(Internet Information Service). Abyss will be the only option
available if you are installing this on a Standard Windows
(Non-Server). IIS is the preferred web server as it is native to the
Windows Server Operating System and other services may already be
using IIS. 

Prior to selecting which Web Server you would prefer to use, please
verify with your network administrators that nothing is already
running on the selected ports or it may cause conflicts and neither
application will work. Abyss will allow you to select different port
numbers such as 5000(HTTP) and 5001(HTTPS), as seen below. Please note
that if you change the ports within Abyss you will not be able to
alter this after the installation. IIS will select the web server
ports which is 80 (HTTP) and 443 (HTTPS) and will not provide an
option to change these ports. If you use IIS please verify that your
HTTP (80) and HTTPS (443) ports are not being used by another
application or service within the same domain. 

In order to use the HTTPS port, a SSL Certificate is needed. SSL
certificates allow secure communication between web servers and
encrypt information to protect data. This encryption is a requirement
on 3CX Web Console. If you already have an SSL Certificate you can
import it - but if you do not already have a certificate you can have
3CX installer generate a certificate. Keep in mind, if you select this
option it will give security warnings to anyone who will access the
webpage. 

If you selected the option to generate a certificate you will be
prompted for you company name as well as a password for your
certificate. Please enter this information and press Next to continue
with the installation 

The next screen will show you all of the information for the
installation as well as give you the option to change any settings by
pressing the Back button. If you are content with your options, you
can select Install and the process can take several minutes depending
on your hardware specifications 

After the installation you will have an icon on your desktop. By
opening this application, it will open a web address in your default
web browser and may take several minutes to load the web page. 

The Icon that is on your desktop should look similar to this: 

After the webpage loads, you will be prompted with a username and a
password which was set up during the installation process. Please
enter the information and login with your credentials 

After you are logged into the web management console you will be
prompted to set up the number of digits for internal extensions. By
default this option is set to 3 digits. Please note that you will not
be able to change this setting after the installation is complete. In
order to change the extension length a fresh install will need to be
performed (uninstalling 3CX and reinstalling). Please note that if you
decide to change the extension from a fresh install, you will not be
able to restore from a backup because the extension length will be
incorrect. 

You will now be prompted with the time zone and country that your 3CX
Phone System is operating from. 

You will now have to set up an operator extension. This extension is
the default destination for inbound calls. Please provide the
extension, voicemail extension, their first name, last name, and email
address. 

After setting up your operator extension you will be able to set up
options for international calls. You can use the arrows on the left
hand side of the region to expand other options and view more specific
regions. By checking additional checkboxes, you will allow outbound
phone calls to be made to other countries. Any checkbox that is not
selected will disallow any external phone calls to be made to that
region, but incoming calls from those regions will still be allowed. 

Your set up is now complete and you should see a screen similar to the
image below - you will also have a quick start guide on the next steps
to complete your 3CX Server. 

 ]]></contentstext>
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<contents><![CDATA[<p>888Voip Engineers have created a Yealink T49G template to Auto-Provision with 3CX. You can find this <a href="http://www.888voip.com/wp-content/uploads/files/yealinkT49_888VoIP.ph.xml" target="_blank">custom template by clicking here.</a> After downloading this template, you will need to save it into two directories in order for 3CX to synchronize and acknowledge that this file exists:</p>
<p>C:\ProgramData\3CX\Instance1\Data\Http\Templates\phones</p>
<p>C:\ProgramData\3CX\Data\Http\Templates\phones</p>
<p><img src="/wp-content/uploads/2016/02/TemplateList1.png" alt="" width="600" /></p>
<p>After saving this ph.xml file into the two directories you will need to restart your 3CX Server in order for these changes to take effect. You will then see the new template under the drop down menus when creating a new extension.</p>
<p><img src="/wp-content/uploads/2016/02/Extension2.png" alt="" width="600" /></p>]]></contents>
<contentstext><![CDATA[888Voip Engineers have created a Yealink T49G template to
Auto-Provision with 3CX. You can find this custom template by clicking
here.
[http://www.888voip.com/wp-content/uploads/files/yealinkT49_888VoIP.ph.xml]
After downloading this template, you will need to save it into two
directories in order for 3CX to synchronize and acknowledge that this
file exists: 

C:ProgramData3CXInstance1DataHttpTemplatesphones 

C:ProgramData3CXDataHttpTemplatesphones 

After saving this ph.xml file into the two directories you will need
to restart your 3CX Server in order for these changes to take effect.
You will then see the new template under the drop down menus when
creating a new extension. 

]]></contentstext>
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<subject><![CDATA[Yealink T49G Template for 3CX Phone System]]></subject>
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<contents><![CDATA[<p><strong>What is provisioning?</strong> Provisioning is the process of registering and configuring a device over the network. Throughout this guide we will explain three methods to provision phones &ndash; Plug and Play, Auto Provision, and Manual Provision. Prior to provisioning a phone using either the Auto Provision or Manual Provision methods you will need to set up an extension within 3CX Web Management Console. You can also do this step prior to Plug and Play or as you are provisioning the device via Plug and Play.</p>
<p>To set up an extension, login to 3CX Web Management Console and select Extensions on the left hand navigation pane. Click Add Extension and fill out information for the extension, passwords, and name &ndash; then apply your settings.</p>
<p>&nbsp;<img src="/wp-content/uploads/2016/02/extensions1.jpg" alt="" width="600" height="151" /></p>
<p><img src="/wp-content/uploads/2016/02/addext2.jpg" alt="" width="600" height="270" /></p>
<p><strong>Plug and Play (PnP)</strong> is the easiest method for provisioning a phone. All of the device information is sent to the PBX using a multicast broadcast. The device will share information such as the MAC Address, IP and Model with 3CX Phone System for easier configuration.</p>
<p>To provision a phone using Plug and Play you need to plug the phone into the network and power it on. You will be able to see the new phone within the 3CX Web Management Console and it will display its information in bold, showing that it is un-provisioned. The phones will be listed under the phones tab on the left hand navigation menu.</p>
<p>Verify that the MAC Address and IP address within 3CX match the phone that you have plugged in and would like to provision, and then you can either select &ldquo;Add Extension&rdquo; or &ldquo;Assign Extension&rdquo;. If you have already created an extension you can Assign the phone to the previously created extension with the option &ldquo;Assign Extension&rdquo;. If you have not created an extension yet, use the option &ldquo;Add Extension&rdquo; to creating the users extension. After assigning the phone to the proper extension, the phone will begin the provisioning process. When the device has downloaded its configuration and completed the boot cycle you have completed the provisioning process.</p>
<p>Please confirm the phone has successfully registered via the Extensions View Tab in 3CX. The &ldquo;registered&rdquo; indicator light for the extension will now be green. (See screenshots below)</p>
<p>&nbsp;<img src="/wp-content/uploads/2016/02/assignext3.jpg" alt="" width="600" height="246" /></p>
<p><img src="/wp-content/uploads/2016/02/selectext4.jpg" alt="" width="460" height="128" /></p>
<p><img src="/wp-content/uploads/2016/02/extview.jpg" alt="" width="600" height="162" /></p>
<p>&nbsp;</p>
<p><strong>Auto-Provisioning</strong> will use the phones web interface to allow you to input the server URL where your configuration files are located. In order to use this process, the devices extension will need to be configured. Once you fill in the users details for the extension, please navigate to the phone provisioning tab. Here you will need to fill in the MAC Address of the device, then select the phone model. Once you apply these settings, a configuration file will be generated and stored in the provisioning web server.</p>
<p><img src="/wp-content/uploads/2016/02/addext6.jpg" alt="" width="600" height="189" /></p>
<p>The URL needed to provision your device can be found on 3CX Web Management Console by clicking &ldquo;System Status&rdquo; and referencing Line #5 (Configure and deploy IP Phones using the Provisioning Link) You can copy and paste this link in the next steps.</p>
<p><img src="/wp-content/uploads/2016/02/systemstatus7.jpg" alt="" width="600" height="267" /></p>
<p>Once you have the URL for provisioning, please enter the Phone&rsquo;s IP Address into your favorite web browser (Firefox, Chrome, Internet Explorer). Please keep in mind that the computer must be on the same local network as the phone in order to connect to the web interface of the phone. You will arrive at a login screen where both the default username and password is set to &ldquo;admin&rdquo; without quotes. After logging into the phone you will be able to navigate to the section for provisioning. On Yealink phones this can be found under Settings &ndash; Auto Provision. Enter the web address where the configuration files will be located (this was found on the previous step). After adding the URL to the phone, click &ldquo;Auto Provision Now&rdquo; and your phone will reboot shortly.</p>
<p><img src="/wp-content/uploads/2016/02/phonelogin8.jpg" alt="" width="306" height="191" /></p>
<p><img src="/wp-content/uploads/2016/02/phoneprovision9.jpg" alt="" width="600" height="261" /></p>
<p>Please confirm the phone has successfully registered via the Extensions View Tab in 3CX. The &ldquo;registered&rdquo; indicator light for the extension will now be green. (See screenshot below)</p>
<p><strong>&nbsp;<img src="/wp-content/uploads/2016/02/extview.jpg" alt="" width="600" height="162" /></strong></p>
<p><strong>Manual provisioning</strong> uses the phones web interface where you will need to manually input the extensions details. Within 3CX you will need to create an extension you would like to use for the new phone before you provision the phone. You also need to know the IP address of the 3CX Phone System as well as the IP address for the phone. Enter the phones IP address into your favorite web browser (Firefox, Chrome, Internet Explorer). The default username and password for the phone web interface is set to &ldquo;admin&rdquo; without quotes<strong>.</strong></p>
<p>After logging into the phone, navigate to Account tab and select &ldquo;Register&rdquo;. You will then need to change the Line Active selection to Enabled (See screenshot below). You will also need to input the following information:</p>
<ul>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Register Name &ndash; Extension</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; User Name &ndash; Extension</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Password &ndash; Password provided while setting up the extension</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Server Host (3CX Server)</li>
</ul>
<p>The settings that will be used for Caller ID and Line Keys are:</p>
<ul>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Label &ndash; This shows on the phones main screen as a line key</li>
<li>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Display Name &ndash; This appears on the caller ID when making internal outbound calls</li>
</ul>
<p>Upon applying these settings your phone will begin to provision and will reboot shortly and the registration status will change from Disabled to Registered.</p>
<p><img src="/wp-content/uploads/2016/02/manualprovision11.jpg" alt="" width="600" height="375" /></p>
<p><img src="/wp-content/uploads/2016/02/manualregister12.jpg" alt="" width="456" height="59" /></p>
<p>Please confirm the phone has successfully registered via the Extensions View Tab in 3CX. The &ldquo;registered&rdquo; indicator light for the extension will now be green. (See screenshot below)</p>
<p>&nbsp;<img src="/wp-content/uploads/2016/02/extview.jpg" alt="" width="600" height="162" /></p>]]></contents>
<contentstext><![CDATA[WHAT IS PROVISIONING? Provisioning is the process of registering and
configuring a device over the network. Throughout this guide we will
explain three methods to provision phones - Plug and Play, Auto
Provision, and Manual Provision. Prior to provisioning a phone using
either the Auto Provision or Manual Provision methods you will need to
set up an extension within 3CX Web Management Console. You can also do
this step prior to Plug and Play or as you are provisioning the device
via Plug and Play. 

To set up an extension, login to 3CX Web Management Console and select
Extensions on the left hand navigation pane. Click Add Extension and
fill out information for the extension, passwords, and name - then
apply your settings. 

PLUG AND PLAY (PNP) is the easiest method for provisioning a phone.
All of the device information is sent to the PBX using a multicast
broadcast. The device will share information such as the MAC Address,
IP and Model with 3CX Phone System for easier configuration. 

To provision a phone using Plug and Play you need to plug the phone
into the network and power it on. You will be able to see the new
phone within the 3CX Web Management Console and it will display its
information in bold, showing that it is un-provisioned. The phones
will be listed under the phones tab on the left hand navigation menu. 

Verify that the MAC Address and IP address within 3CX match the phone
that you have plugged in and would like to provision, and then you can
either select "Add Extension" or "Assign Extension". If you have
already created an extension you can Assign the phone to the
previously created extension with the option "Assign Extension". If
you have not created an extension yet, use the option "Add Extension"
to creating the users extension. After assigning the phone to the
proper extension, the phone will begin the provisioning process. When
the device has downloaded its configuration and completed the boot
cycle you have completed the provisioning process. 

Please confirm the phone has successfully registered via the
Extensions View Tab in 3CX. The "registered" indicator light for the
extension will now be green. (See screenshots below) 

AUTO-PROVISIONING will use the phones web interface to allow you to
input the server URL where your configuration files are located. In
order to use this process, the devices extension will need to be
configured. Once you fill in the users details for the extension,
please navigate to the phone provisioning tab. Here you will need to
fill in the MAC Address of the device, then select the phone model.
Once you apply these settings, a configuration file will be generated
and stored in the provisioning web server. 

The URL needed to provision your device can be found on 3CX Web
Management Console by clicking "System Status" and referencing Line #5
(Configure and deploy IP Phones using the Provisioning Link) You can
copy and paste this link in the next steps. 

Once you have the URL for provisioning, please enter the Phone's IP
Address into your favorite web browser (Firefox, Chrome, Internet
Explorer). Please keep in mind that the computer must be on the same
local network as the phone in order to connect to the web interface of
the phone. You will arrive at a login screen where both the default
username and password is set to "admin" without quotes. After logging
into the phone you will be able to navigate to the section for
provisioning. On Yealink phones this can be found under Settings -
Auto Provision. Enter the web address where the configuration files
will be located (this was found on the previous step). After adding
the URL to the phone, click "Auto Provision Now" and your phone will
reboot shortly. 

Please confirm the phone has successfully registered via the
Extensions View Tab in 3CX. The "registered" indicator light for the
extension will now be green. (See screenshot below) 

MANUAL PROVISIONING uses the phones web interface where you will need
to manually input the extensions details. Within 3CX you will need to
create an extension you would like to use for the new phone before you
provision the phone. You also need to know the IP address of the 3CX
Phone System as well as the IP address for the phone. Enter the phones
IP address into your favorite web browser (Firefox, Chrome, Internet
Explorer). The default username and password for the phone web
interface is set to "admin" without quotes. 

After logging into the phone, navigate to Account tab and select
"Register". You will then need to change the Line Active selection to
Enabled (See screenshot below). You will also need to input the
following information: 

 	* Register Name - Extension
 	* User Name - Extension
 	* Password - Password provided while setting up the extension
 	* Server Host (3CX Server)

The settings that will be used for Caller ID and Line Keys are: 

 	* Label - This shows on the phones main screen as a line key
 	* Display Name - This appears on the caller ID when making internal
outbound calls

Upon applying these settings your phone will begin to provision and
will reboot shortly and the registration status will change from
Disabled to Registered. 

Please confirm the phone has successfully registered via the
Extensions View Tab in 3CX. The "registered" indicator light for the
extension will now be green. (See screenshot below) 

 ]]></contentstext>
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<subject><![CDATA[Provisioning Yealink Phones With 3CX]]></subject>
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<contents><![CDATA[<p><strong><span style="font-size: 12pt;">Overview:</span>&nbsp;</strong>Also known as a Recovery Mode Update, the TFTP process can be used to save what can be seen as an otherwise dead phone. This process is most useful in the scenario that a phone either gets stuck in a boot loop, or stays on a Welcome Initializing...Please Wait screen. However, it can also be useful on a device where a factory reset does not seem to be resetting the device correctly. If you find that&nbsp;your device fits&nbsp;into any&nbsp;of these categories, &nbsp;the TFTP recovery can be performed;&nbsp;it&nbsp;will reflash the firmware on the device in the case of a corrupt firmware&nbsp;and if successful, will allow your device to fully boot again.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink1.jpg" alt="Welcome Screen" width="600" height="420" /></p>
<p><span style="font-size: 12pt;"><strong>Set-up Guide:</strong></span> In order to proceed with this process and be able to successfully perform the TFTP recovery, you are going to need:</p>
<ul>
<li>We will be using the&nbsp;<a href="http://kin.klever.net/pumpkin/binaries">PumpKIN&nbsp;TFTP Server</a> in this guide but you can substitute this with a TFTP program&nbsp;of your choice, however the guide will not be covering how to configure those.</li>
<li>PC with web access</li>
<li>Phone on the same network as that PC.</li>
</ul>
<p>&nbsp;</p>
<p><strong><span style="font-size: 12pt;">Step 1: Gathering the Necessary Files</span></strong></p>
<p>You will be able to find all of the files necessary for the recovery of your device on our <a href="https://drive.google.com/folderview?id=0By_uJpOr3h2YVTl4Z2JoOFpGMjA&amp;usp=sharing">shared google drive</a>.</p>
<p>From here, you can navigate to the folder for the device that you will be trying to recover, inside of that folder you will find all of the files necessary to perform the TFTP recovery on that device. Download these files and place them in a folder that you will have easy access to. Make sure that you have downloaded all the files in your devices folder, the process will not be successful if you have not acquired all of the files necessary for your device.</p>
<p>&nbsp;</p>
<p><span style="font-size: 12pt;"><strong>Step 2:&nbsp;</strong><strong>Setting Up Your TFTP Server</strong></span></p>
<p>The process to setup your TFTP server is not overly difficult. Once downloaded from the link above, allow PumpKIN to install and then open it once it is installed. You will see a screen as below.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/pumpkin11.jpg" alt="Pumpkin Main Screen" width="600" height="379" /></p>
<p>Now, you are going to want to hit the options button and you will be presented with a screen as shown below.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/pumpkin21.jpg" alt="" width="600" height="463" /></p>
<p>On this screen, you are going to want to set the TFTP filesystem root to the folder that you downloaded the files from above in, in this examples case, they are located in a TFTP folder that is on my desktop. You will also want to ensure that you change the read request behavior to give all files and the write request behavior to take all files, so that way the TFTP program will know to allow the phone to request all of the files it is looking for in that folder. Once this is done you can click OK to save your changes and you are ready to move on to the next step.</p>
<p><span style="font-size: 12pt;"><strong>Step 3: TFTP Recovery</strong></span></p>
<p>Depending upon what device you are going to be recovering, this process will deviate slightly.</p>
<ul>
<li><span style="font-size: 12pt;"><strong>T4x, T3x, T2x, T19 and CP860</strong></span></li>
</ul>
<p style="padding-left: 30px;">&nbsp;</p>
<p style="padding-left: 30px;"><strong>Step 1:&nbsp;</strong>With your device off, press and hold down on the speakerphone button. While holding this button down, either plug in the power adapter or reinsert your PoE cable. Keep holding this button down until you are presented with a screen as shown below.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink21.jpg" alt="" width="600" height="400" /></p>
<p>&nbsp;</p>
<p style="padding-left: 60px;"><strong>***Please Note***&nbsp;</strong>The <a href="http://www.888voip.com/products/yealink-cp860.html">CP860</a> follows a slightly different process, instead of holding down the speakerphone button during this process, you will be holding down the second soft key button from the left as there is no speakerphone button to hold down on the <a href="http://www.888voip.com/products/yealink-cp860.html">CP860</a>. This is circled in the picture below.</p>
<p style="padding-left: 60px;"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink4.jpg" alt="" width="600" height="514" /></p>
<p style="padding-left: 60px;">&nbsp;</p>
<p style="padding-left: 30px;"><strong>Step 2:&nbsp;</strong>On this screen you will see 4 options, IP Addr:, NetMask:, GateWay:, and TFTP IP: as pictured below. For the IP Addr:, you are going to want to insert an unused IP address that is available on your network, in our example we are using 192.168.90.236. Next, for the NetMask:, you are going to want to insert the subnet mask that your network is operating under, in our example we are using 255.255.255.0. However, this can be different depending upon your network configuration. Next, for the GateWay:, you are going to want to insert the IP address that your router is located at, in our example that would be 192.168.90.1. Finally, you are going to want to insert the IP address of your PC that is hosting PumpKIN for the TFTP IP:. Our TFTP server is located on a PC at IP 192.168.90.3 so we inserted that value for our TFTP IP:. Now, in the case that you are not aware of your network details, please contact your network administrator and they can help you get that information. Once you have filled these details out, press the OK key on the phone and it will start requesting the files that you downloaded from earlier.&nbsp;</p>
<p style="padding-left: 30px;"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink3.jpg" alt="Detail Screen" width="400" height="185" /></p>
<p style="padding-left: 30px;"><strong>Step 3:&nbsp;</strong>The process can take several minutes to complete and during that time you will see a ...Updating...Please Wait displayed on the screen. Allow it to complete it's process, and the phone will reboot once it is complete. If it was a successful flash, you will be presented with a factory reset device that has been recovered. If you are presented with an Update Fail... screen you will want to ensure that the details that you entered on the phone are correct and that you have pointed PumpKIN at the correct folder. Also, please ensure that you have downloaded the correct files and all of the files necessary for your device.</p>
<p style="padding-left: 30px;">&nbsp;</p>
<p style="padding-left: 30px;">&nbsp;</p>
<ul>
<li><a href="http://www.888voip.com/products/yealink-w52p.html"><span style="font-size: 12pt;"><strong>W52P</strong></span></a></li>
</ul>
<p style="padding-left: 30px;">&nbsp;</p>
<p style="padding-left: 30px;"><strong>Step 1:&nbsp;</strong>The process to recover a W52P base unit is slightly different when compared to recovering a phone. First, you will need to get the IP address information on your PC that will be hosting PumpKIN as configured below, these are the default values that the W52P will look for when put into recovery mode.</p>
<p style="padding-left: 30px;"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/ipdetails.jpg" alt="" width="448" height="504" /></p>
<p style="padding-left: 30px;">&nbsp;</p>
<p style="padding-left: 30px;"><strong>Step 2:&nbsp;</strong>During this process the base unit will get an IP address of 192.168.0.100 and will look to the IP address as configured above for its recovery files. While the unit is off, long press the button on the front of the device, it looks like a small dot with three waves eminating out from the center. While continuing to hold this button reconnect your power adapter or PoE cable. Continue holding until you see the three lights on the base unit turn on one by one (power led --&gt; network led --&gt; phone led)</p>
<p style="padding-left: 30px;">&nbsp;</p>
<p style="padding-left: 30px;"><strong>Step 3:&nbsp;</strong>This process can take several minutes to complete, however once sucessful the unit will reboot and once you have solid lights on the front of the device you will be able to access the device at the IP address that it has received from DHCP.</p>
<p style="padding-left: 30px;">&nbsp;</p>
<p>Once the steps above have been completed sucessfully, your device&nbsp;will have rebooted and you will have sucessfully recovered your device.&nbsp;</p>]]></contents>
<contentstext><![CDATA[OVERVIEW:Also known as a Recovery Mode Update, the TFTP process can be
used to save what can be seen as an otherwise dead phone. This process
is most useful in the scenario that a phone either gets stuck in a
boot loop, or stays on a Welcome Initializing...Please Wait screen.
However, it can also be useful on a device where a factory reset does
not seem to be resetting the device correctly. If you find that your
device fits into any of these categories, the TFTP recovery can be
performed; it will reflash the firmware on the device in the case of a
corrupt firmware and if successful, will allow your device to fully
boot again. 

SET-UP GUIDE: In order to proceed with this process and be able to
successfully perform the TFTP recovery, you are going to need: 

 	* We will be using the PumpKIN TFTP Server
[http://kin.klever.net/pumpkin/binaries] in this guide but you can
substitute this with a TFTP program of your choice, however the guide
will not be covering how to configure those.
 	* PC with web access
 	* Phone on the same network as that PC.

STEP 1: GATHERING THE NECESSARY FILES 

You will be able to find all of the files necessary for the recovery
of your device on our shared google drive
[https://drive.google.com/folderview?id=0By_uJpOr3h2YVTl4Z2JoOFpGMjA&usp=sharing].


From here, you can navigate to the folder for the device that you will
be trying to recover, inside of that folder you will find all of the
files necessary to perform the TFTP recovery on that device. Download
these files and place them in a folder that you will have easy access
to. Make sure that you have downloaded all the files in your devices
folder, the process will not be successful if you have not acquired
all of the files necessary for your device. 

STEP 2:SETTING UP YOUR TFTP SERVER 

The process to setup your TFTP server is not overly difficult. Once
downloaded from the link above, allow PumpKIN to install and then open
it once it is installed. You will see a screen as below. 

Now, you are going to want to hit the options button and you will be
presented with a screen as shown below. 

On this screen, you are going to want to set the TFTP filesystem root
to the folder that you downloaded the files from above in, in this
examples case, they are located in a TFTP folder that is on my
desktop. You will also want to ensure that you change the read request
behavior to give all files and the write request behavior to take all
files, so that way the TFTP program will know to allow the phone to
request all of the files it is looking for in that folder. Once this
is done you can click OK to save your changes and you are ready to
move on to the next step. 

STEP 3: TFTP RECOVERY 

Depending upon what device you are going to be recovering, this
process will deviate slightly. 

 	* T4X, T3X, T2X, T19 AND CP860

STEP 1:With your device off, press and hold down on the speakerphone
button. While holding this button down, either plug in the power
adapter or reinsert your PoE cable. Keep holding this button down
until you are presented with a screen as shown below. 

***PLEASE NOTE***The CP860
[http://www.888voip.com/products/yealink-cp860.html] follows a
slightly different process, instead of holding down the speakerphone
button during this process, you will be holding down the second soft
key button from the left as there is no speakerphone button to hold
down on the CP860
[http://www.888voip.com/products/yealink-cp860.html]. This is circled
in the picture below. 

STEP 2:On this screen you will see 4 options, IP Addr:, NetMask:,
GateWay:, and TFTP IP: as pictured below. For the IP Addr:, you are
going to want to insert an unused IP address that is available on your
network, in our example we are using 192.168.90.236. Next, for the
NetMask:, you are going to want to insert the subnet mask that your
network is operating under, in our example we are using 255.255.255.0.
However, this can be different depending upon your network
configuration. Next, for the GateWay:, you are going to want to insert
the IP address that your router is located at, in our example that
would be 192.168.90.1. Finally, you are going to want to insert the IP
address of your PC that is hosting PumpKIN for the TFTP IP:. Our TFTP
server is located on a PC at IP 192.168.90.3 so we inserted that value
for our TFTP IP:. Now, in the case that you are not aware of your
network details, please contact your network administrator and they
can help you get that information. Once you have filled these details
out, press the OK key on the phone and it will start requesting the
files that you downloaded from earlier. 

STEP 3:The process can take several minutes to complete and during
that time you will see a ...Updating...Please Wait displayed on the
screen. Allow it to complete it's process, and the phone will reboot
once it is complete. If it was a successful flash, you will be
presented with a factory reset device that has been recovered. If you
are presented with an Update Fail... screen you will want to ensure
that the details that you entered on the phone are correct and that
you have pointed PumpKIN at the correct folder. Also, please ensure
that you have downloaded the correct files and all of the files
necessary for your device. 

 	* W52P [http://www.888voip.com/products/yealink-w52p.html]

STEP 1:The process to recover a W52P base unit is slightly different
when compared to recovering a phone. First, you will need to get the
IP address information on your PC that will be hosting PumpKIN as
configured below, these are the default values that the W52P will look
for when put into recovery mode. 

STEP 2:During this process the base unit will get an IP address of
192.168.0.100 and will look to the IP address as configured above for
its recovery files. While the unit is off, long press the button on
the front of the device, it looks like a small dot with three waves
eminating out from the center. While continuing to hold this button
reconnect your power adapter or PoE cable. Continue holding until you
see the three lights on the base unit turn on one by one (power led
--> network led --> phone led) 

STEP 3:This process can take several minutes to complete, however once
sucessful the unit will reboot and once you have solid lights on the
front of the device you will be able to access the device at the IP
address that it has received from DHCP. 

Once the steps above have been completed sucessfully, your device will
have rebooted and you will have sucessfully recovered your device. ]]></contentstext>
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<subject><![CDATA[How to use the TFTP Recovery Mode on Yealink Devices]]></subject>
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<contents><![CDATA[<p><strong>What is Alert-Info and&nbsp;how can it be used for paging with Polycom IP Phones?</strong></p>
<p><br />When present in an INVITE request, the Alert-Info header field can activate features like an alternative ring tone for User Agents, or have a SIP phone auto-answer a call.</p>
<p>This blog details the necessary settings needed to make your <a href="http://www.888voip.com/products/polycom-phones/">Polycom IP Phones</a> automatically pick up a page placed to it, with no user interaction. You can make your SIP phones auto answer by setting the Alert-Info variable in your PBX and modifying the XML configuration file.</p>
<p>Typically, all Polycom phones will retrieve their configuration files from a server on bootup. It is on this server where you will need to make modifications to the configuration files. These changes can be made in the 'sip.cfg' file located on the provisioning server or a customized configuration file used for provisioning. The filename is customizable so you may need to ask the Polycom phone administrator for its location and then possibly request the following changes are made.</p>
<p>&nbsp;Below is a modified config which will enable paging to Polycom phones with the ability to set the phone to auto-answer with ringer or without ringer.</p>
<p>&nbsp;</p>
<p>&nbsp;&lt;?xml version="1.0" encoding="UTF-8" standalone="yes"?&gt;</p>
<p>&lt;!-- PlcmConversionCreatedFile version=1.2 converted=Tue Aug 23 16:59:18 2011 --&gt;</p>
<p>&lt;polycomConfig xmlns:xsi="<a href="http://www.w3.org/2001/XMLSchema-instance">http://www.w3.org/2001/XMLSchema-instance</a>" xsi:noNamespaceSchemaLocation="polycomConfig.xsd"&gt;</p>
<p>&lt;voIpProt&gt;</p>
<p>&lt;voIpProt.SIP&gt;</p>
<p><span style="background-color: #ffff00;">&lt;voIpProt.SIP.alertInfo voIpProt.SIP.alertInfo.1.class="ringAutoAnswer" voIpProt.SIP.alertInfo.1.value="Auto Answer"/&gt; voIpProt.SIP.alertInfo.2.class="autoAnswer" voIpProt.SIP.alertInfo.2.value="answer"</span></p>
<p>&lt;/voIpProt.SIP&gt;</p>
<p>&lt;/voIpProt&gt;</p>
<p>&lt;/polycomConfig&gt;</p>
<p>&nbsp;</p>
<p>You will then need your PBX to send the following in the INVITE.<br />&nbsp;</p>
<p>Alert-Info: info=Auto Answer - This will ring then answer.</p>
<p>Alert-Info: info=Answer - This will just auto answer.</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p>&nbsp;</p>]]></contents>
<contentstext><![CDATA[WHAT IS ALERT-INFO ANDHOW CAN IT BE USED FOR PAGING WITH POLYCOM IP
PHONES? 

When present in an INVITE request, the Alert-Info header field can
activate features like an alternative ring tone for User Agents, or
have a SIP phone auto-answer a call. 

This blog details the necessary settings needed to make your Polycom
IP Phones [http://www.888voip.com/products/polycom-phones/]
automatically pick up a page placed to it, with no user interaction.
You can make your SIP phones auto answer by setting the Alert-Info
variable in your PBX and modifying the XML configuration file. 

Typically, all Polycom phones will retrieve their configuration files
from a server on bootup. It is on this server where you will need to
make modifications to the configuration files. These changes can be
made in the 'sip.cfg' file located on the provisioning server or a
customized configuration file used for provisioning. The filename is
customizable so you may need to ask the Polycom phone administrator
for its location and then possibly request the following changes are
made. 

 Below is a modified config which will enable paging to Polycom phones
with the ability to set the phone to auto-answer with ringer or
without ringer. 

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

<!-- PlcmConversionCreatedFile version=1.2 converted=Tue Aug 23
16:59:18 2011 --> 

<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance
[http://www.w3.org/2001/XMLSchema-instance]"
xsi:noNamespaceSchemaLocation="polycomConfig.xsd"> 

<voIpProt> 

<voIpProt.SIP> 

<voIpProt.SIP.alertInfo
voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
voIpProt.SIP.alertInfo.1.value="Auto Answer"/>
voIpProt.SIP.alertInfo.2.class="autoAnswer"
voIpProt.SIP.alertInfo.2.value="answer" 

</voIpProt.SIP> 

</voIpProt> 

</polycomConfig> 

You will then need your PBX to send the following in the INVITE.

Alert-Info: info=Auto Answer - This will ring then answer. 

Alert-Info: info=Answer - This will just auto answer. 

 ]]></contentstext>
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<subject><![CDATA[Paging with Polycom Phones using Alert-Info.]]></subject>
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<contents><![CDATA[<p>In this blog we will detail the common methods that paging is delivered over an IP network and discuss how these methods are deployed over a network infrastructure such as an IP PBX like 3CX or Asterisk, paging endpoints, IP Phones for desktop paging, overhead speakers, and gateways that are SIP enabled for large&nbsp;paging solutions. Companies like&nbsp;<a href="http://www.888voip.com/products/cyberdata/">Cyberdata</a>&nbsp;have SIP based overhead ceiling speakers, SIP based paging amplifiers, paging gateways and loudspeakers with horns to accompany them.</p>
<p><strong>UNICAST IP PAGING</strong></p>
<p>To understand how IP paging works, we will look at the two types of paging solutions that are supported on most IP based PBX&rsquo;s. The first is called &ldquo;unicast&rdquo;, or more commonly, a&nbsp;unicast page. Unicast pages are delivered on a one to one basis and the IP PBX is the source for this page. In most common unicast paging setups, administrators will setup a single paging group or multiple page groups in the IP PBX configuration. For our example, we&rsquo;ll use a paging ring group assigned to extension 400.</p>
<p>Paging group 400 then contains SIP extensions that belong to individual users on the system and these extensions are physical SIP endpoints such as an IP&nbsp;Phone. In unicast paging environments, when a user dials paging group 400 from their phone, the PBX sends SIP and RTP audio traffic to and from each one of the&nbsp;endpoints in the ring group. Once the PBX connects to the endpoints, they will auto-answer and the caller can begin speaking. In a 3CX environment this type of page&nbsp;would only use 1 simultaneous call to the ring group, as the audio traffic is one way to the endpoints in the ring group. On some PBX's, and in the example below, each endpoint has a connection established to it from the device that is paging, thus taxing the PBX's CPU and processing power. 3CX handles this differently and does not burden the PBX when paging to ring group.</p>
<p>&nbsp;</p>
<p><strong><img src="/wp-content/uploads/2016/04/Unicast-Paging-Functions-Diagram1.jpg" alt="" width="608" height="240" /></strong></p>
<p><strong>MULTICAST IP PAGING</strong></p>
<p>Multicast paging achieves the same outcome as unicast paging but delivers each page much differently. In most multi-cast paging environments, administrators will specify a single multi-cast paging IP address in the IP PBX setup. They will then configure each paging endpoint such as an IP Phone, overhead speaker, or amplifier, to&nbsp;&ldquo;listen in&rdquo; on that multicast addresses. The IP Phones or other paging endpoints are always &ldquo;listening&rdquo; to this address and when RTP packets or audio is heard, the phone or paging endpoints simply play that audio stream.</p>
<p>In both of these paging environments, two types of paging can be performed. Desktop paging and overhead paging. We will talk about these two options and offer&nbsp;product suggestions based on these examples.</p>
<p><img src="/wp-content/uploads/2016/04/Multi-Cast-Paging-Diagram1.jpg" alt="" width="608" height="239" /></p>
<p><strong>DESKTOP PAGING</strong></p>
<p>Desktop paging is exactly what it sounds like, paging at the desktop. This involves an IP Phone that supports paging. During a desktop page, the IP PBX will usually&nbsp;be configured for a page group or groups depending on the scenario, and a user will simply dial into that page group to initiate the desktop page. The SIP extensions&nbsp;of the users inside that page group in the IP PBX are then paged either via a unicast or multicast stream.</p>
<p>Desktop pages are delivered directly to the IP Phone, triggering it to immediately go off-hook and present the page of the speakerphone. So, from a hardware&nbsp;standpoint, you need an IP Phone that supports a speakerphone, which is the vast majority of them on the market today. If you are unsure, please don&rsquo;t hesitate to&nbsp;contact us and speak with one of our sales representatives.</p>
<p>Desktop paging can also be configured to &ldquo;play a tone&rdquo; first before the page comes through the speaker to alert the user that a page is being delivered. Also, pages&nbsp;can be configured to interrupt a user on a call or wait until the user has completed the call to page through. In cases where a page will interrupt a user during a phone&nbsp;conversation, the party that this user is speaking with will not be able to hear the page; however this may cause a distraction to all users on the conversation.</p>
<p><strong>OVERHEAD PAGING</strong></p>
<p>The second type of IP Paging is overhead paging. Great advancements have been made in this department over the past few years to cater to new IP based overhead&nbsp;paging solutions. Most of these products are powered via POE so installation is easily accomplished with 1 cable. Configuration is very easy as most of these endpoints are&nbsp;based upon SIP standards and are easily configured with a SIP extension (just like you would with an IP Phone) via the products web GUI configuration. Simply add&nbsp;these endpoints as SIP extensions to a new or existing page group on your SIP PBX and dial the page group extension.</p>
<p><img src="/wp-content/uploads/2016/04/IP-Paging-Typical-Installation.jpg" alt="" width="589" height="227" /></p>
<p><strong>EXISTING ANALOG PAGING SYSTEMS</strong><br /> <a href="http://www.888voip.com/products/cyberdata-paging-gateway.html"><strong>Cyberdata Paging Gateway</strong></a><br /> You may come across a client who has converted their phone system to an IP PBX but they want to incorporate their existing analog paging system to their new VoIP&nbsp;phone system. Cyberdata offers legacy based devices that will help address this need. The first is the Cyberdata Paging Gateway. Like any gateway does in the IP&nbsp;world, it integrates 2 different networks together. Most commonly, this is done over a voice circuit where voice gateways connect to standard RJ-11 analog POTS lines&nbsp;or analog telephones and allow communication to an IP network or IP based PBX. The same is true for the Cyberdata Paging gateway however this gateway is used to&nbsp;connect older legacy analog paging systems to an IP network. When users dial the page group extension, the IP PBX will ring the SIP extension registered to the&nbsp;paging gateway and it will send the page to the analog paging equipment attached to it.</p>
<p><a href="http://www.888voip.com/products/cyberdata-voip-paging-zone-controller.html"><strong>Cyberdata Zone Controller</strong></a><br /> The second device used in the paging world to incorporate older analog paging systems with new IP based PBX&rsquo;s is the Cyberdata Zone Controller. Unlike the paging&nbsp;gateway, the zone controller contains (4) standard audio out ports which are used to connect to existing analog amplifiers. In turn, speakers or horns would then be&nbsp;connected to those amplifiers.</p>
<p>Once the standard audio connection is made between the zone controller and existing amplifiers, the zone controller will communicate to the IP PBX via its LAN port&nbsp;over the IP network. The zone controller contains 4 page zones and allows for up to 15 zone groups.</p>
<p>&nbsp;</p>]]></contents>
<contentstext><![CDATA[In this blog we will detail the common methods that paging is
delivered over an IP network and discuss how these methods are
deployed over a network infrastructure such as an IP PBX like 3CX or
Asterisk, paging endpoints, IP Phones for desktop paging, overhead
speakers, and gateways that are SIP enabled for large paging
solutions. Companies like Cyberdata
[http://www.888voip.com/products/cyberdata/] have SIP based overhead
ceiling speakers, SIP based paging amplifiers, paging gateways and
loudspeakers with horns to accompany them. 

UNICAST IP PAGING 

To understand how IP paging works, we will look at the two types of
paging solutions that are supported on most IP based PBX's. The first
is called "unicast", or more commonly, a unicast page. Unicast pages
are delivered on a one to one basis and the IP PBX is the source for
this page. In most common unicast paging setups, administrators will
setup a single paging group or multiple page groups in the IP PBX
configuration. For our example, we'll use a paging ring group assigned
to extension 400. 

Paging group 400 then contains SIP extensions that belong to
individual users on the system and these extensions are physical SIP
endpoints such as an IP Phone. In unicast paging environments, when a
user dials paging group 400 from their phone, the PBX sends SIP and
RTP audio traffic to and from each one of the endpoints in the ring
group. Once the PBX connects to the endpoints, they will auto-answer
and the caller can begin speaking. In a 3CX environment this type of
page would only use 1 simultaneous call to the ring group, as the
audio traffic is one way to the endpoints in the ring group. On some
PBX's, and in the example below, each endpoint has a connection
established to it from the device that is paging, thus taxing the
PBX's CPU and processing power. 3CX handles this differently and does
not burden the PBX when paging to ring group. 

MULTICAST IP PAGING 

Multicast paging achieves the same outcome as unicast paging but
delivers each page much differently. In most multi-cast paging
environments, administrators will specify a single multi-cast paging
IP address in the IP PBX setup. They will then configure each paging
endpoint such as an IP Phone, overhead speaker, or amplifier, to
"listen in" on that multicast addresses. The IP Phones or other paging
endpoints are always "listening" to this address and when RTP packets
or audio is heard, the phone or paging endpoints simply play that
audio stream. 

In both of these paging environments, two types of paging can be
performed. Desktop paging and overhead paging. We will talk about
these two options and offer product suggestions based on these
examples. 

DESKTOP PAGING 

Desktop paging is exactly what it sounds like, paging at the desktop.
This involves an IP Phone that supports paging. During a desktop page,
the IP PBX will usually be configured for a page group or groups
depending on the scenario, and a user will simply dial into that page
group to initiate the desktop page. The SIP extensions of the users
inside that page group in the IP PBX are then paged either via a
unicast or multicast stream. 

Desktop pages are delivered directly to the IP Phone, triggering it to
immediately go off-hook and present the page of the speakerphone. So,
from a hardware standpoint, you need an IP Phone that supports a
speakerphone, which is the vast majority of them on the market today.
If you are unsure, please don't hesitate to contact us and speak with
one of our sales representatives. 

Desktop paging can also be configured to "play a tone" first before
the page comes through the speaker to alert the user that a page is
being delivered. Also, pages can be configured to interrupt a user on
a call or wait until the user has completed the call to page through.
In cases where a page will interrupt a user during a phone
conversation, the party that this user is speaking with will not be
able to hear the page; however this may cause a distraction to all
users on the conversation. 

OVERHEAD PAGING 

The second type of IP Paging is overhead paging. Great advancements
have been made in this department over the past few years to cater to
new IP based overhead paging solutions. Most of these products are
powered via POE so installation is easily accomplished with 1 cable.
Configuration is very easy as most of these endpoints are based upon
SIP standards and are easily configured with a SIP extension (just
like you would with an IP Phone) via the products web GUI
configuration. Simply add these endpoints as SIP extensions to a new
or existing page group on your SIP PBX and dial the page group
extension. 

EXISTING ANALOG PAGING SYSTEMS
 CYBERDATA PAGING GATEWAY
[http://www.888voip.com/products/cyberdata-paging-gateway.html]
 You may come across a client who has converted their phone system to
an IP PBX but they want to incorporate their existing analog paging
system to their new VoIP phone system. Cyberdata offers legacy based
devices that will help address this need. The first is the Cyberdata
Paging Gateway. Like any gateway does in the IP world, it integrates 2
different networks together. Most commonly, this is done over a voice
circuit where voice gateways connect to standard RJ-11 analog POTS
lines or analog telephones and allow communication to an IP network or
IP based PBX. The same is true for the Cyberdata Paging gateway
however this gateway is used to connect older legacy analog paging
systems to an IP network. When users dial the page group extension,
the IP PBX will ring the SIP extension registered to the paging
gateway and it will send the page to the analog paging equipment
attached to it. 

CYBERDATA ZONE CONTROLLER
[http://www.888voip.com/products/cyberdata-voip-paging-zone-controller.html]
 The second device used in the paging world to incorporate older
analog paging systems with new IP based PBX's is the Cyberdata Zone
Controller. Unlike the paging gateway, the zone controller contains
(4) standard audio out ports which are used to connect to existing
analog amplifiers. In turn, speakers or horns would then be connected
to those amplifiers. 

Once the standard audio connection is made between the zone controller
and existing amplifiers, the zone controller will communicate to the
IP PBX via its LAN port over the IP network. The zone controller
contains 4 page zones and allows for up to 15 zone groups. 

 ]]></contentstext>
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<author />
<email />
<subject><![CDATA[Designing and implementing a VoIP Paging System.]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong>&nbsp;</p>
<p>Included with <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> is the 3CX Tunnel, also known as the 3CX SBC. This software, which can be run on either a <a href="http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html">Raspberry Pi</a> or a Windows PC, allows for easier connection of remote extensions to a remote 3CX phone system. Using port 5090 by default, (unless being utilized in a multi-tenant array). The 3CX Tunnel combines all SIP and RTP VoIP Packets from one location and can deliver them to the&nbsp;3CX servers location on a single port. This allows the traffic to&nbsp;help&nbsp;overcome various firewall and mobile network&nbsp;issues and allows a simpler route to configuring remote extensions on your system. Some of the other reasons to use it&nbsp;include:</p>
<ul>
<li>Resolve issues of NAT Traversal at the remote location.</li>
<li>Simplified Firewall configuration at both the remote and the PBX location.</li>
<li>Overcome difficulties with ISP's that block VoIP Traffic based on port numbers.</li>
<li>Allows VoIP-over-WiFi in some restricted locations</li>
<li>Can help overcome troublesome firewalls that can not handle VoIP traffic correctly<br /> or are problematic to configure.</li>
</ul>
<p>&nbsp;</p>
<p><strong>Usage:</strong>&nbsp;</p>
<p>The 3CX Tunnel functions using port 5090 by default, unless being utilized in a multi-tenant environment, and helps allow easier remote connection of extensions back to the 3CX server. In order to do this, you will need to setup the 3CX SBC software on either a Windows PC, or a <a href="http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html">Raspberry Pi</a>. (the 3CX tunnel will be installed at the phones location, not on the PBX's LAN) The Windows SBC can support up to 16 phones, and the <a href="http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html">Raspberry Pi</a> can support up to 5 phones. Using more on either device can lead to dropped calls, so it is not recommended to go over that limit. However, you can have multiple SBC's running on one LAN, splitting up the devices. When using the 3CX tunnel you will only have to worry about forwarding the designated tunnel port.&nbsp;</p>
<p>One use case for the 3CX SBC; say you have two or more phones located at a remote site, the SBC will route the signaling back to the phone system through the port selected, however it will keep the media streams local for extension to extension calls at the remote site. This will help limit bandwidth requirements for phones at the remote site.</p>
<p>Another usage scenario for the 3CX Tunnel&nbsp;is for remote 3CXPhone users. The 3CXPhone application has a built in tunnel which when configured at the extension level, will automatically be used when it is detected that it is not on the LAN. This will help voip calls navigate on mobile networks where port 5060 can be blocked by the mobile network provider.&nbsp;</p>
<p><strong>Configuration:</strong></p>
<p>There are several steps needed to ensure that your system is correctly setup to use the 3CX tunnel.</p>
<p>First, let's configure the PBX.</p>
<p>In the 3CX Management Console, look under settings for the Security --&gt; 3CX Tunnel tab. The tunnel password is set to a randomly generated value, you can use this password or change it to something different. You will also see the local IP and tunnel port options. These do not need to be changed and in a cloud system cannot be changed. &nbsp;The tunnel port on a single install of 3cx will default to 5090, in a multi-tenant cloud system,&nbsp;this will change depending upon which tenant you are. For example, Instance 9 on a multi-tenant server will default to&nbsp;tunnel port 13090.</p>
<p>Click OK and the tunnel service will restart automatically.</p>
<p>Next, we are going to look at the configuration of the firewall. When using the 3CX tunnel or 3CX SBC, you will only need to open up the designated tunnel port for remote connections. Please remember to open both UDP and TCP connections through this port.</p>
<p><strong>Configuration on Remote Sites:</strong></p>
<p>Prior to installing the 3CX SBC software, you will want to ensure that all other network adapters on your PC are disabled outside of the one you are going to be using, this can include, Bluetooth, Wireless, and additional Ethernet ports. You will also want to disable IPv6 on the network adapter that you are going to be using. If you do not take these steps, the SBC software can conflict with the different network adapters and lead to 3CX pulling an incorrect IP address for the location of your SBC. If you change this in 3CX to the correct IP your phones will provision, however, it is likely that any outbound calls from that phone will give you a 403 Forbidden error. An example of this faulty IP address is shown below. In our case it is showing the SBC at 169.254.199.132:5060 which is not correct. Our SBC is located at 192.168.90.175.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/wrongip.png" alt="" width="600" height="103" /></p>
<p>The 3CX SBC software can be installed on either a Windows PC or a <a href="http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html">Raspberry Pi</a>. We will go over how to install the software on a Windows PC in this guide.</p>
<p>First, you will need to download the <a href="http://downloads.3cx.com/downloads/3CXSBC14.msi">3CX SBC for Windows software</a>. Go ahead and launch the installer once it has completed downloading.</p>
<p>After you get past the terms &amp; conditions you will be faced with this screen.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/3cx1.jpg" alt="" width="573" height="432" /></p>
<p>You will enter the details from your 3CX server here. In this example, our FQDN is testpbx.test.com. This will vary&nbsp;depending upon your configuration and can either be your public/external IP or the FQDN. In a single install, you should not need to change the ports here, however, if you're in a multi-tenant environment,&nbsp;these values will be different depending upon which tenant you are. Based on the&nbsp;earlier example, Tenant 9 would need you&nbsp;to change the 3CX PBX SIP Port to 13060 and the 3CX Tunnel Port to 13090.</p>
<p>On the next page you will see the option for a failover PBX. If have another server on standby, you can enable this option and enter the public IP of that 3CX server and the SBC software will know to roll-over to that server when your primary server has failed.</p>
<p>Finally, you will be presented with the screen below.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/3cx2.jpg" alt="" width="546" height="435" /></p>
<p>You are going to enter the password that was generated by 3CX for you in the earlier steps,&nbsp;unless you changed it to something different. You also have the ability to turn encryption on or off. By default it is on and it adds a layer of security by encrypting all of the calls passing through the 3CX Tunnel. Go ahead and click next and the 3CX SBC service will install and start. You can now move onto configuring phones to work through the 3CX SBC.</p>
<p><strong>Configuration of a Phone on the SBC:</strong></p>
<p>We will go over the manual and automatic configuration of a <a href="http://www.888voip.com/featured-solution/yealink/">Yealink</a> device to the 3CX SBC in this article.</p>
<p>To begin, create an extension on <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a>, for this example we will be using 203. Once completed, you will want to return to the phone's gui. Go ahead and enter the details from <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> for the extension you created, (Register Name, User Name, Password). Proceed to the&nbsp;SIP Server settings, this will be set to the FQDN of your hosted server, or the public IP of that server depending upon your setup. In this example, this is a hosted server that is positioned as tenant 9, and&nbsp;is using port 13060 to communicate. The port will vary depending upon your setup.&nbsp;Finally, you will need to set your outbound proxy settings. This will&nbsp;be set to the IP address of the Windows PC/<a href="http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html">Raspberry Pi</a> that you installed the SBC&nbsp;software&nbsp;on. In this example, it&nbsp;is located at 192.168.90.175 and the phones will still communicate back to the SBC using port 5060. It is just the connection from the SBC to the <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> server that is using the designated tunnel port. These settings are pictured below. Apply the settings and you should see the device register.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealinkops.jpg" alt="" width="600" height="519" /></p>
<p>&nbsp;On <a href="http://www.888voip.com/featured-solution/yealink/">Yealink</a> and other supported devices, you can also auto-provision through the SBC with <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a>.</p>
<p>&nbsp;To do this, proceed to the&nbsp;phones tab in <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a>. <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> will recognize all the phones that are plugged into the network that the SBC is setup on. This will allow you to easily autoprovision and set phones up through the 3CX management console. The image below is how&nbsp;unprovisioned phones on the SBC's network will show up in 3CX.</p>
<p>&nbsp;<img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink11.jpg" alt="" width="600" height="161" /></p>
<p>&nbsp;</p>
<p>Select the phone you would like&nbsp;to provision and assign the device to an existing extension, or create a new extension. When assigning a phone to an existing or new extension, 3CX will automatically fill in the necessary information. You will want to ensure that the private IP address of the SBC is located in the IP address of 3CX SBC field. Once completed and you have applied the changes, the phone will receive the NOTIFY&nbsp;to provision and will pull down it's config from the 3CX server. This page is pictured below.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/yealink2.jpg" alt="" width="600" height="355" /></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: 

Included with 3CX [http://www.888voip.com/featured-solution/3cx/] is
the 3CX Tunnel, also known as the 3CX SBC. This software, which can be
run on either a Raspberry Pi
[http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html] or
a Windows PC, allows for easier connection of remote extensions to a
remote 3CX phone system. Using port 5090 by default, (unless being
utilized in a multi-tenant array). The 3CX Tunnel combines all SIP and
RTP VoIP Packets from one location and can deliver them to the 3CX
servers location on a single port. This allows the traffic to help
overcome various firewall and mobile network issues and allows a
simpler route to configuring remote extensions on your system. Some of
the other reasons to use it include: 

 	* Resolve issues of NAT Traversal at the remote location.
 	* Simplified Firewall configuration at both the remote and the PBX
location.
 	* Overcome difficulties with ISP's that block VoIP Traffic based on
port numbers.
 	* Allows VoIP-over-WiFi in some restricted locations

	* Can help overcome troublesome firewalls that can not handle VoIP
traffic correctly
 or are problematic to configure. 

USAGE: 

The 3CX Tunnel functions using port 5090 by default, unless being
utilized in a multi-tenant environment, and helps allow easier remote
connection of extensions back to the 3CX server. In order to do this,
you will need to setup the 3CX SBC software on either a Windows PC, or
a Raspberry Pi
[http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html].
(the 3CX tunnel will be installed at the phones location, not on the
PBX's LAN) The Windows SBC can support up to 16 phones, and the
Raspberry Pi
[http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html]
can support up to 5 phones. Using more on either device can lead to
dropped calls, so it is not recommended to go over that limit.
However, you can have multiple SBC's running on one LAN, splitting up
the devices. When using the 3CX tunnel you will only have to worry
about forwarding the designated tunnel port. 

One use case for the 3CX SBC; say you have two or more phones located
at a remote site, the SBC will route the signaling back to the phone
system through the port selected, however it will keep the media
streams local for extension to extension calls at the remote site.
This will help limit bandwidth requirements for phones at the remote
site. 

Another usage scenario for the 3CX Tunnel is for remote 3CXPhone
users. The 3CXPhone application has a built in tunnel which when
configured at the extension level, will automatically be used when it
is detected that it is not on the LAN. This will help voip calls
navigate on mobile networks where port 5060 can be blocked by the
mobile network provider. 

CONFIGURATION: 

There are several steps needed to ensure that your system is correctly
setup to use the 3CX tunnel. 

First, let's configure the PBX. 

In the 3CX Management Console, look under settings for the Security
--> 3CX Tunnel tab. The tunnel password is set to a randomly generated
value, you can use this password or change it to something different.
You will also see the local IP and tunnel port options. These do not
need to be changed and in a cloud system cannot be changed. The tunnel
port on a single install of 3cx will default to 5090, in a
multi-tenant cloud system, this will change depending upon which
tenant you are. For example, Instance 9 on a multi-tenant server will
default to tunnel port 13090. 

Click OK and the tunnel service will restart automatically. 

Next, we are going to look at the configuration of the firewall. When
using the 3CX tunnel or 3CX SBC, you will only need to open up the
designated tunnel port for remote connections. Please remember to open
both UDP and TCP connections through this port. 

CONFIGURATION ON REMOTE SITES: 

Prior to installing the 3CX SBC software, you will want to ensure that
all other network adapters on your PC are disabled outside of the one
you are going to be using, this can include, Bluetooth, Wireless, and
additional Ethernet ports. You will also want to disable IPv6 on the
network adapter that you are going to be using. If you do not take
these steps, the SBC software can conflict with the different network
adapters and lead to 3CX pulling an incorrect IP address for the
location of your SBC. If you change this in 3CX to the correct IP your
phones will provision, however, it is likely that any outbound calls
from that phone will give you a 403 Forbidden error. An example of
this faulty IP address is shown below. In our case it is showing the
SBC at 169.254.199.132:5060 which is not correct. Our SBC is located
at 192.168.90.175. 

The 3CX SBC software can be installed on either a Windows PC or a
Raspberry Pi
[http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html].
We will go over how to install the software on a Windows PC in this
guide. 

First, you will need to download the 3CX SBC for Windows software
[http://downloads.3cx.com/downloads/3CXSBC14.msi]. Go ahead and launch
the installer once it has completed downloading. 

After you get past the terms & conditions you will be faced with this
screen. 

You will enter the details from your 3CX server here. In this example,
our FQDN is testpbx.test.com. This will vary depending upon your
configuration and can either be your public/external IP or the FQDN.
In a single install, you should not need to change the ports here,
however, if you're in a multi-tenant environment, these values will be
different depending upon which tenant you are. Based on the earlier
example, Tenant 9 would need you to change the 3CX PBX SIP Port to
13060 and the 3CX Tunnel Port to 13090. 

On the next page you will see the option for a failover PBX. If have
another server on standby, you can enable this option and enter the
public IP of that 3CX server and the SBC software will know to
roll-over to that server when your primary server has failed. 

Finally, you will be presented with the screen below. 

You are going to enter the password that was generated by 3CX for you
in the earlier steps, unless you changed it to something different.
You also have the ability to turn encryption on or off. By default it
is on and it adds a layer of security by encrypting all of the calls
passing through the 3CX Tunnel. Go ahead and click next and the 3CX
SBC service will install and start. You can now move onto configuring
phones to work through the 3CX SBC. 

CONFIGURATION OF A PHONE ON THE SBC: 

We will go over the manual and automatic configuration of a Yealink
[http://www.888voip.com/featured-solution/yealink/] device to the 3CX
SBC in this article. 

To begin, create an extension on 3CX
[http://www.888voip.com/featured-solution/3cx/], for this example we
will be using 203. Once completed, you will want to return to the
phone's gui. Go ahead and enter the details from 3CX
[http://www.888voip.com/featured-solution/3cx/] for the extension you
created, (Register Name, User Name, Password). Proceed to the SIP
Server settings, this will be set to the FQDN of your hosted server,
or the public IP of that server depending upon your setup. In this
example, this is a hosted server that is positioned as tenant 9, and
is using port 13060 to communicate. The port will vary depending upon
your setup. Finally, you will need to set your outbound proxy
settings. This will be set to the IP address of the Windows
PC/Raspberry Pi
[http://www.888voip.com/products/888voip-3cx-raspberry-pi-sbc.html]
that you installed the SBC software on. In this example, it is located
at 192.168.90.175 and the phones will still communicate back to the
SBC using port 5060. It is just the connection from the SBC to the 3CX
[http://www.888voip.com/featured-solution/3cx/] server that is using
the designated tunnel port. These settings are pictured below. Apply
the settings and you should see the device register. 

 On Yealink [http://www.888voip.com/featured-solution/yealink/] and
other supported devices, you can also auto-provision through the SBC
with 3CX [http://www.888voip.com/featured-solution/3cx/]. 

 To do this, proceed to the phones tab in 3CX
[http://www.888voip.com/featured-solution/3cx/]. 3CX
[http://www.888voip.com/featured-solution/3cx/] will recognize all the
phones that are plugged into the network that the SBC is setup on.
This will allow you to easily autoprovision and set phones up through
the 3CX management console. The image below is how unprovisioned
phones on the SBC's network will show up in 3CX. 

Select the phone you would like to provision and assign the device to
an existing extension, or create a new extension. When assigning a
phone to an existing or new extension, 3CX will automatically fill in
the necessary information. You will want to ensure that the private IP
address of the SBC is located in the IP address of 3CX SBC field. Once
completed and you have applied the changes, the phone will receive the
NOTIFY to provision and will pull down it's config from the 3CX
server. This page is pictured below. 

]]></contentstext>
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<subject><![CDATA[Benefits and Configuration of the 3CX Tunnel/3CX SBC]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong></p>
<p>In this article, we will be covering the configuration of several models of Welltech Gateways with the <a href="http://www.888voip.com/featured-solution/3cx/">3CX Phone System</a>. <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> natively supports&nbsp;3 different variations of Welltech Gateways, the <a href="#ATA172">ATA 172 Plus PoE</a>, the <a href="#2424s">Wellgate 2424s</a> and the the <a href="#2540">Wellgate 2540</a>. Depending upon which gateway you choose, the configuration will&nbsp;differ slightly as outlined below.</p>
<p>&nbsp;</p>
<p id="ATA172"><a href="http://www.888voip.com/products/welltech-ata172plus-poe.html"><span style="font-size: 14pt;"><strong>Welltech ATA 172 Plus PoE:</strong></span></a></p>
<p><strong>Step 1: Preparing <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> and the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">Welltech ATA 172 Plus</a></strong></p>
<p>Before setting up the ATA&nbsp;you will want&nbsp;to ensure that the device is factory reset and that you have updated your device to the <a href="http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/welltechata.ssh">latest supported firmware that is available</a>.</p>
<ol>
<li>To factory reset the device, look on the back for a Reset button, you will need to hold this in for 5 seconds. The device will power cycle and reset it's configuration back to factory default.</li>
<li>You will need to figure out what IP address that this device has taken through DHCP unless you have set a MAC reservation for the device. There are two ways to easily find the IP address, first, we use a program known as <a href="http://www.filehippo.com/download_advanced_ip_scanner/?utm_source=FT&amp;utm_medium=Redirect&amp;utm_campaign=AIS">Advanced IP Scanner</a>, this will scan your network for the devices located in the same network, and you can search by MAC address to find which IP your Welltech ATA has taken. The other way is to connect an analog phone to one of the FXS ports and dial "#126#", this will tell you what IP address the device has received.</li>
<li>Open a web browser and type in the IP address with port 9999 attached at the end. For example, our ATA is located at <a href="http://192.168.90.127:9999">http://192.168.90.127:9999</a>. This will take you to the management console where you will need to log in using the default username and password, which are <strong>root</strong> and <strong>test</strong> respectively.</li>
<li>Navigate to the management tab in the web interface and locate the Firmware tab within. From here, you will need to browse to the firmware that you downloaded above and then select Upgrade. This will begin the firmware update process on the device. It will take between 2 and 3 minutes to complete and be sure to not turn off power to the device during this process.</li>
<li>Once complete you can log back into the web interface using the details from above and you are ready for the configuration of the device.</li>
</ol>
<p>&nbsp;</p>
<p><strong>Step 2: Adding the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">Welltech ATA 172 Plus</a> to <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a></strong></p>
<ol>
<li>Before continuing, you will want to proceed to the 3CX Management Console, from here select the Updates tab and navigate to the VoIP Gateway template updates. Ensure that you have the latest template available for the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">ATA 172 Plus</a> and if you do not, select it and update it.</li>
<li>Proceed to the extensions tab and create two new extensions that you plan on using with the ATA. For our example, we will be using 500 and 501.</li>
<li>Go to&nbsp;the VoIP/PSTN Gateways tab, select the "Add Gateway" option. Enter a name for the gateway and select "Welltech" from the drop down list. Finally select the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">ATA172plus-PoE</a> as the model of gateway and click next.</li>
<li>On the tone set selection, select the country that this device is going to be used in. Enter the devices MAC address, and select your method for faxing if it is different from the default, click next when done.</li>
<li>On this page you will need to enter the IP address that the gateway is located at or that you are going to have it located at either statically or through a DHCP reservation, for this example that is 192.168.90.127.</li>
<li>Proceed to the next page, it will now be asking you to select the extensions that you will want to use with the Welltech ATA, we will be using extensions 500 and 501 that we created earlier.</li>
<li>Finally, proceed to the final page and click the "Generate config file" button. This will create a mac.dat file that is going to be used to provision the ATA. You are going to want to save or move this file to the following directory:</li>
</ol>
<p style="padding-left: 90px;">C:\ProgramData\3CX\Instance1\<wbr />Data\Http\Interface\<wbr />provisioning\PROVSUBDIR\</p>
<p style="padding-left: 90px;">&nbsp;</p>
<p><strong>Step 3: Provisioning the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">Welltech ATA172plus-PoE</a></strong></p>
<ol>
<li>&nbsp;Return to the web interface for the <a href="http://www.888voip.com/products/welltech-ata172plus-poe.html">Welltech ATA172plus-PoE</a> and go to the Management --&gt; Auto Provision tab.</li>
<li>&nbsp;You are now going to want to enter your server details now, this will be pictured below.</li>
<li>&nbsp;To provision with 3CX you are going to want to select HTTP as the "Provision Active:". In the HTTP server, you will enter your FQDN or the IP of your server depending upon your setup. The HTTP port will be either 80 if using IIS, or 5000 if using Abyss.</li>
<li>&nbsp;Finally, the file path is going to be from the end of the provisioning link that you can find on your system status screen.</li>
</ol>
<p><a href="/wp-content/uploads/2016/04/ata172.jpg"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/04/ata172.jpg" alt="" width="600" height="269" /></a>Submit these changes and click "save and reboot" on the next screen, this will reboot your device and it will pull down it's configuration on this reboot. Once completed, you will see the extensions appear as registered in the <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> management console.</p>
<p>&nbsp;</p>
<p id="2424s"><span style="font-size: 14pt;"><strong><a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Welltech Wellgate 2424s:</a></strong></span></p>
<p>&nbsp;<strong>Step 1: Preparing <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> and the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a></strong></p>
<p>Before setting up the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a> FXS VoIP gateway, you will want&nbsp;to ensure that the device is factory reset and has been updated to the <a href="http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/wg24.4.02.bin">latest supported firmware that is available</a>.</p>
<ol>
<li>If you are unsure that your <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a> is at factory default settings, press and hold the button on the back of the device for 5 seconds to factory reset it, the device will power cycle and be back to factory default settings.</li>
<li>Connect your PC to the LAN port on the <a href="Wellgate%202424s">Wellgate 2424s</a>.</li>
<li>In order to access the GUI of the device, you will need to configure your PC to have a static IP of 192.168.123.111.</li>
<li>On your web browser, go to 192.168.123.123, this is the default LAN IP of the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a>. To login, you will use the default username and password,<strong> root</strong> and <strong>root</strong> respectively</li>
<li>Navigate to the Maintenance tab and choose firmware update from within, browse to the file that you downloaded earlier and press import to begin the firmware update.</li>
<li>This process will take 2 to 3 minutes to complete, be sure to not power off the device before the update is complete.</li>
<li>Once finished, you can log back into the web interface using the details from above and you are ready for the configuration of the device.</li>
</ol>
<p>&nbsp;</p>
<p>&nbsp;<strong>Step 2: Adding the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a> to <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a></strong></p>
<ol>
<li>Before continuing, you will want to proceed to the <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> Management Console, from here select the Updates tab and navigate to the VoIP Gateway template updates. Ensure that you have the latest template available for the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a>&nbsp;and if you do not, select it and update it.</li>
<li>Proceed to the extensions tab and create the&nbsp;extensions that you plan on using with the FXS gateway. The 2424s is a 24-Port FXS Gateway and can support up to 24 extensions.</li>
<li>Select the VoIP/PSTN Gateways tab and select the "Add Gateway" option within. Enter a name for the gateway and select "Welltech" from the Brand: drop down list. Finally under the Model: drop down, select the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Wellgate 2424s</a>. Click next when completed.</li>
<li>Select the country the device will be used in, and enter the devices WAN MAC address, click next.</li>
<li>Now it will be asking for the IP address that the device will be located at, it is best to have reserved an option in your DHCP server so the IP does not change and you lose access to the devices that it is controlling.</li>
<li>Proceed to the next tab, this will be asking you to select the extensions that you are going to be using with the FXS ports. Once you have completed this, proceed to the final&nbsp;page and generate the config file. This file will need to be saved or moved to&nbsp;this directory.</li>
</ol>
<p style="text-align: left; padding-left: 90px;">C:\ProgramData\3CX\Instance1\<wbr />Data\Http\Interface\<wbr />provisioning\PROVSUBDIR\</p>
<p style="text-align: left; padding-left: 90px;">&nbsp;</p>
<p style="text-align: left; padding-left: 90px;">&nbsp;</p>
<p style="text-align: left;"><strong>Step 3: Provisioning the <a href="http://www.888voip.com/products/welltech-wellgate-2424s.html">Welltech&nbsp;Wellgate 2424s</a></strong></p>
<ol>
<li style="text-align: left;">Return to the web interface and log back in using the credentials outlined earlier in this article.</li>
<li style="text-align: left;">To provision this device we will be using the WAN port rather then the LAN port and will have to get this configured to work correctly.</li>
<li style="text-align: left;">Proceed to the the Device Setting --&gt; Network tab. You will be presented with a screen as below,
<ul>
<li style="text-align: left;"><a href="/wp-content/uploads/2016/05/2424_1.png" target="_blank"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/2424_1.jpg" alt="" width="600" height="219" /></a></li>
</ul>
</li>
<li style="text-align: left;">You are going to change the Network Type to DHCP and DNS Server Mode to Auto if you have a DHCP reservation in your router. Go ahead and apply the changes.
<ul>
<li style="text-align: left;"><a href="/wp-content/uploads/2016/05/2424_2.png" target="_blank"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/2424_2.jpg" alt="" width="600" height="216" /></a></li>
</ul>
</li>
<li style="text-align: left;">Reboot the gateway and switch the gateway to the network it will be located on, ensure that you have plugged into the WAN port.</li>
<li style="text-align: left;">If you have not setup a MAC reservation for this device you will have to locate the IP that it is on using a program such as Advanced IP Scanner. Proceed to this IP and login to the web interface using the details from above.</li>
<li style="text-align: left;">Proceed to the Device Setting --&gt; Provisioning tab and configure as follows:
<ul>
<li style="text-align: left;">Provisioning Type --&gt; HTTP</li>
<li style="text-align: left;">HTTP Config URL --&gt; <a href="http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR">http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR</a></li>
<li style="text-align: left;">Refresh Interval (minute) --&gt; 14400</li>
<li style="text-align: left;"><a href="/wp-content/uploads/2016/05/2540_2small.jpg" target="_blank"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/2540_2small.jpg" alt="" width="600" height="194" /></a></li>
</ul>
</li>
<li style="text-align: left;">Apply these settings and proceed to the Maintenance tab, choose reboot and apply your changes.</li>
<li style="text-align: left;">On the reboot the device will download it's configuration from the 3CX server and your device will be able to make and receive calls when complete. The extensions will appear Green in the extension status tab if the device took its configuration correctly.</li>
</ol>
<p>&nbsp;</p>
<p>&nbsp;</p>
<p id="2540"><a href="http://www.888voip.com/products/welltech-wellgate-2540.html"><span style="font-size: 14pt;"><strong>Welltech Wellgate 2540:</strong></span></a></p>
<p><strong>Step 1: Preparing <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a> and the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate&nbsp;2540</a></strong></p>
<p>Before setting up the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a>&nbsp;gateway, it is best to ensure that the device is factory reset and has been updated to the&nbsp;<a href="http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/wg25.4.02.bin">latest supported firmware that is available</a>.</p>
<ol>
<li>If you are unsure that your <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a> is at factory default settings, press and hold the button on the back of the device for 5 seconds to factory reset it.</li>
<li>Connect your PC to the LAN port on the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a>.</li>
<li>In order to access the GUI of the device, you will need to configure your PC to a static IP of 192.168.123.111.</li>
<li>On your web browser, go to 192.168.123.123, this is the default LAN IP of the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a>. To login, you will use the default username and password,<strong>&nbsp;root</strong>&nbsp;and&nbsp;<strong>root</strong>&nbsp;respectively</li>
<li>Navigate to the Maintenance tab and choose firmware update from within, browse to the file that you downloaded earlier and press import to begin the firmware update.</li>
<li>This process will take 2 to 3 minutes to complete, be sure to not power off the device before the update is complete.</li>
<li>Once complete you can log back into the web interface using the details from above and you are ready for the configuration of the device.</li>
</ol>
<p>&nbsp;</p>
<p>&nbsp;<strong>Step 2: Adding the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a> to <a href="http://www.888voip.com/featured-solution/3cx/">3CX</a></strong></p>
<ol>
<li>Before continuing, you will want to proceed to the 3CX Management Console, from here select the Updates tab and navigate to the VoIP Gateway template updates. Ensure that you have the latest template available for the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Wellgate 2540</a>&nbsp;and if you do not, select it and update it.</li>
<li>Proceed to the VoIP/PSTN Gateways tab on the 3CX Management Console. Select Add Gateway, provide a name for this gateway and select Welltech from the drop down list. Finally ensure that you have the 2540 selected before clicking next.</li>
<li>Enter the Welltech's WAN MAC address on this screen. Make sure it is the WAN MAC and not the LAN MAC.</li>
<li>On the next page you will be entering the IP Address of the Welltech device. This will either need to be statically set or have a DHCP reservation set for the Welltech device as you do not want the IP changing on your gateway.</li>
<li>The next page will be the port creation page, you will need to configure your inbound routes on this page. Click next</li>
<li>Here you will be needed to create your outbound routes for the Welltech device.</li>
<li>Finally proceed to the final page, go ahead and generate the configuration file. This will need to be either saved or moved into the directory below before your device will be able to sucessfully provision.</li>
</ol>
<p style="padding-left: 90px;">C:\ProgramData\3CX\Instance1\<wbr />Data\Http\Interface\<wbr />provisioning\PROVSUBDIR\</p>
<p style="padding-left: 90px;">&nbsp;</p>
<p><strong>Step 3: Provisioning the <a href="http://www.888voip.com/products/welltech-wellgate-2540.html">Welltech&nbsp;Wellgate 2540</a></strong></p>
<ol>
<li>Return to the web interface and log back in using the credentials outlined earlier in this article.</li>
<li>To provision this device we will be using the WAN port rather then the LAN port and will have to get this configured to work correctly.</li>
<li>Proceed to the the Device Setting --&gt; Network tab. You are going to change the Network Type to DHCP and DNS Server Mode to Auto. This will be pictured below. Go ahead and apply the changes.
<ul>
<li><a href="/wp-content/uploads/2016/05/2540_1.png" target="_blank"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/2540_1.jpg" alt="" width="600" height="243" /></a></li>
</ul>
</li>
<li>Reboot the gateway and switch the gateway to the network it will be located on, ensure that you have plugged into the WAN port.</li>
<li>If you have not setup a MAC reservation for this device you will have to locate the IP that it is on using a program such as Advanced IP Scanner. Proceed to this IP and login to the web interface using the details from above.</li>
<li>Proceed to the Device Setting --&gt; Provisioning tab and configure as follows:
<ul>
<li>Provisioning Type --&gt; HTTP</li>
<li>HTTP Config URL --&gt;&nbsp;<a href="http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR">http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR</a></li>
<li>Refresh Interval (minute) --&gt; 14400</li>
<li><a href="/wp-content/uploads/2016/05/2540_21.jpg" target="_blank"><img style="display: block; margin-left: auto; margin-right: auto;" src="/wp-content/uploads/2016/05/2540_2small.jpg" alt="" width="600" height="194" /></a></li>
</ul>
</li>
<li>Apply these settings and proceed to the Maintenance tab, choose reboot and apply your changes.</li>
<li>On the reboot the device will download it's configuration from the 3CX server and your device will be able to make and receive calls when complete. The ports&nbsp;will appear Green in the ports/trunks&nbsp;status tab if the device took its configuration correctly.</li>
</ol>
<p>&nbsp;</p>]]></contents>
<contentstext><![CDATA[OVERVIEW: 

In this article, we will be covering the configuration of several
models of Welltech Gateways with the 3CX Phone System
[http://www.888voip.com/featured-solution/3cx/]. 3CX
[http://www.888voip.com/featured-solution/3cx/] natively supports 3
different variations of Welltech Gateways, the ATA 172 Plus PoE, the
Wellgate 2424s and the the Wellgate 2540. Depending upon which gateway
you choose, the configuration will differ slightly as outlined below. 

WELLTECH ATA 172 PLUS POE:
[http://www.888voip.com/products/welltech-ata172plus-poe.html] 

STEP 1: PREPARING 3CX [HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/]
AND THE WELLTECH ATA 172 PLUS
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-ATA172PLUS-POE.HTML] 

Before setting up the ATA you will want to ensure that the device is
factory reset and that you have updated your device to the latest
supported firmware that is available
[http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/welltechata.ssh].


 	* To factory reset the device, look on the back for a Reset button,
you will need to hold this in for 5 seconds. The device will power
cycle and reset it's configuration back to factory default.
 	* You will need to figure out what IP address that this device has
taken through DHCP unless you have set a MAC reservation for the
device. There are two ways to easily find the IP address, first, we
use a program known as Advanced IP Scanner
[http://www.filehippo.com/download_advanced_ip_scanner/?utm_source=FT&utm_medium=Redirect&utm_campaign=AIS],
this will scan your network for the devices located in the same
network, and you can search by MAC address to find which IP your
Welltech ATA has taken. The other way is to connect an analog phone to
one of the FXS ports and dial "#126#", this will tell you what IP
address the device has received.
 	* Open a web browser and type in the IP address with port 9999
attached at the end. For example, our ATA is located at
http://192.168.90.127:9999 [http://192.168.90.127:9999]. This will
take you to the management console where you will need to log in using
the default username and password, which are ROOT and TEST
respectively.
 	* Navigate to the management tab in the web interface and locate the
Firmware tab within. From here, you will need to browse to the
firmware that you downloaded above and then select Upgrade. This will
begin the firmware update process on the device. It will take between
2 and 3 minutes to complete and be sure to not turn off power to the
device during this process.
 	* Once complete you can log back into the web interface using the
details from above and you are ready for the configuration of the
device.

STEP 2: ADDING THE WELLTECH ATA 172 PLUS
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-ATA172PLUS-POE.HTML] TO 3CX
[HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/] 

 	* Before continuing, you will want to proceed to the 3CX Management
Console, from here select the Updates tab and navigate to the VoIP
Gateway template updates. Ensure that you have the latest template
available for the ATA 172 Plus
[http://www.888voip.com/products/welltech-ata172plus-poe.html] and if
you do not, select it and update it.
 	* Proceed to the extensions tab and create two new extensions that
you plan on using with the ATA. For our example, we will be using 500
and 501.
 	* Go to the VoIP/PSTN Gateways tab, select the "Add Gateway" option.
Enter a name for the gateway and select "Welltech" from the drop down
list. Finally select the ATA172plus-PoE
[http://www.888voip.com/products/welltech-ata172plus-poe.html] as the
model of gateway and click next.
 	* On the tone set selection, select the country that this device is
going to be used in. Enter the devices MAC address, and select your
method for faxing if it is different from the default, click next when
done.
 	* On this page you will need to enter the IP address that the
gateway is located at or that you are going to have it located at
either statically or through a DHCP reservation, for this example that
is 192.168.90.127.
 	* Proceed to the next page, it will now be asking you to select the
extensions that you will want to use with the Welltech ATA, we will be
using extensions 500 and 501 that we created earlier.
 	* Finally, proceed to the final page and click the "Generate config
file" button. This will create a mac.dat file that is going to be used
to provision the ATA. You are going to want to save or move this file
to the following directory:

C:ProgramData3CXInstance1DataHttpInterfaceprovisioningPROVSUBDIR 

STEP 3: PROVISIONING THE WELLTECH ATA172PLUS-POE
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-ATA172PLUS-POE.HTML] 

 	* Return to the web interface for the Welltech ATA172plus-PoE
[http://www.888voip.com/products/welltech-ata172plus-poe.html] and go
to the Management --> Auto Provision tab.
 	* You are now going to want to enter your server details now, this
will be pictured below.
 	* To provision with 3CX you are going to want to select HTTP as the
"Provision Active:". In the HTTP server, you will enter your FQDN or
the IP of your server depending upon your setup. The HTTP port will be
either 80 if using IIS, or 5000 if using Abyss.
 	* Finally, the file path is going to be from the end of the
provisioning link that you can find on your system status screen.

 [http://www.888voip.com/wp-content/uploads/2016/04/ata172.jpg]Submit
these changes and click "save and reboot" on the next screen, this
will reboot your device and it will pull down it's configuration on
this reboot. Once completed, you will see the extensions appear as
registered in the 3CX [http://www.888voip.com/featured-solution/3cx/]
management console. 

WELLTECH WELLGATE 2424S:
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2424S.HTML] 

 STEP 1: PREPARING 3CX [HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/]
AND THE WELLGATE 2424S
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2424S.HTML] 

Before setting up the Wellgate 2424s
[http://www.888voip.com/products/welltech-wellgate-2424s.html] FXS
VoIP gateway, you will want to ensure that the device is factory reset
and has been updated to the latest supported firmware that is
available
[http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/wg24.4.02.bin].


 	* If you are unsure that your Wellgate 2424s
[http://www.888voip.com/products/welltech-wellgate-2424s.html] is at
factory default settings, press and hold the button on the back of the
device for 5 seconds to factory reset it, the device will power cycle
and be back to factory default settings.
 	* Connect your PC to the LAN port on the Wellgate 2424s
[http://www.888voip.com/Wellgate%202424s].
 	* In order to access the GUI of the device, you will need to
configure your PC to have a static IP of 192.168.123.111.
 	* On your web browser, go to 192.168.123.123, this is the default
LAN IP of the Wellgate 2424s
[http://www.888voip.com/products/welltech-wellgate-2424s.html]. To
login, you will use the default username and password, ROOT and ROOT
respectively
 	* Navigate to the Maintenance tab and choose firmware update from
within, browse to the file that you downloaded earlier and press
import to begin the firmware update.
 	* This process will take 2 to 3 minutes to complete, be sure to not
power off the device before the update is complete.
 	* Once finished, you can log back into the web interface using the
details from above and you are ready for the configuration of the
device.

 STEP 2: ADDING THE WELLGATE 2424S
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2424S.HTML] TO 3CX
[HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/] 

 	* Before continuing, you will want to proceed to the 3CX
[http://www.888voip.com/featured-solution/3cx/] Management Console,
from here select the Updates tab and navigate to the VoIP Gateway
template updates. Ensure that you have the latest template available
for the Wellgate 2424s
[http://www.888voip.com/products/welltech-wellgate-2424s.html] and if
you do not, select it and update it.
 	* Proceed to the extensions tab and create the extensions that you
plan on using with the FXS gateway. The 2424s is a 24-Port FXS Gateway
and can support up to 24 extensions.
 	* Select the VoIP/PSTN Gateways tab and select the "Add Gateway"
option within. Enter a name for the gateway and select "Welltech" from
the Brand: drop down list. Finally under the Model: drop down, select
the Wellgate 2424s
[http://www.888voip.com/products/welltech-wellgate-2424s.html]. Click
next when completed.
 	* Select the country the device will be used in, and enter the
devices WAN MAC address, click next.
 	* Now it will be asking for the IP address that the device will be
located at, it is best to have reserved an option in your DHCP server
so the IP does not change and you lose access to the devices that it
is controlling.
 	* Proceed to the next tab, this will be asking you to select the
extensions that you are going to be using with the FXS ports. Once you
have completed this, proceed to the final page and generate the config
file. This file will need to be saved or moved to this directory.

C:ProgramData3CXInstance1DataHttpInterfaceprovisioningPROVSUBDIR 

STEP 3: PROVISIONING THE WELLTECHWELLGATE 2424S
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2424S.HTML] 

 	* Return to the web interface and log back in using the credentials
outlined earlier in this article.
 	* To provision this device we will be using the WAN port rather then
the LAN port and will have to get this configured to work correctly.

	* Proceed to the the Device Setting --> Network tab. You will be
presented with a screen as below, 

 	* [http://www.888voip.com/wp-content/uploads/2016/05/2424_1.png]

	* You are going to change the Network Type to DHCP and DNS Server
Mode to Auto if you have a DHCP reservation in your router. Go ahead
and apply the changes. 

 	* [http://www.888voip.com/wp-content/uploads/2016/05/2424_2.png]

 	* Reboot the gateway and switch the gateway to the network it will
be located on, ensure that you have plugged into the WAN port.
 	* If you have not setup a MAC reservation for this device you will
have to locate the IP that it is on using a program such as Advanced
IP Scanner. Proceed to this IP and login to the web interface using
the details from above.

	* Proceed to the Device Setting --> Provisioning tab and configure as
follows: 

 	* Provisioning Type --> HTTP
 	* HTTP Config URL -->
http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR
[http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR]
 	* Refresh Interval (minute) --> 14400
 	*
[http://www.888voip.com/wp-content/uploads/2016/05/2540_2small.jpg]

 	* Apply these settings and proceed to the Maintenance tab, choose
reboot and apply your changes.
 	* On the reboot the device will download it's configuration from the
3CX server and your device will be able to make and receive calls when
complete. The extensions will appear Green in the extension status tab
if the device took its configuration correctly.

WELLTECH WELLGATE 2540:
[http://www.888voip.com/products/welltech-wellgate-2540.html] 

STEP 1: PREPARING 3CX [HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/]
AND THE WELLGATE2540
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2540.HTML] 

Before setting up the Wellgate 2540
[http://www.888voip.com/products/welltech-wellgate-2540.html] gateway,
it is best to ensure that the device is factory reset and has been
updated to the latest supported firmware that is available
[http://downloads.3cx.com/downloads/misc/phonefirmware/Welltech/wg25.4.02.bin].


 	* If you are unsure that your Wellgate 2540
[http://www.888voip.com/products/welltech-wellgate-2540.html] is at
factory default settings, press and hold the button on the back of the
device for 5 seconds to factory reset it.
 	* Connect your PC to the LAN port on the Wellgate 2540
[http://www.888voip.com/products/welltech-wellgate-2540.html].
 	* In order to access the GUI of the device, you will need to
configure your PC to a static IP of 192.168.123.111.
 	* On your web browser, go to 192.168.123.123, this is the default
LAN IP of the Wellgate 2540
[http://www.888voip.com/products/welltech-wellgate-2540.html]. To
login, you will use the default username and password,ROOT and ROOT
respectively
 	* Navigate to the Maintenance tab and choose firmware update from
within, browse to the file that you downloaded earlier and press
import to begin the firmware update.
 	* This process will take 2 to 3 minutes to complete, be sure to not
power off the device before the update is complete.
 	* Once complete you can log back into the web interface using the
details from above and you are ready for the configuration of the
device.

 STEP 2: ADDING THE WELLGATE 2540
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2540.HTML] TO 3CX
[HTTP://WWW.888VOIP.COM/FEATURED-SOLUTION/3CX/] 

 	* Before continuing, you will want to proceed to the 3CX Management
Console, from here select the Updates tab and navigate to the VoIP
Gateway template updates. Ensure that you have the latest template
available for the Wellgate 2540
[http://www.888voip.com/products/welltech-wellgate-2540.html] and if
you do not, select it and update it.
 	* Proceed to the VoIP/PSTN Gateways tab on the 3CX Management
Console. Select Add Gateway, provide a name for this gateway and
select Welltech from the drop down list. Finally ensure that you have
the 2540 selected before clicking next.
 	* Enter the Welltech's WAN MAC address on this screen. Make sure it
is the WAN MAC and not the LAN MAC.
 	* On the next page you will be entering the IP Address of the
Welltech device. This will either need to be statically set or have a
DHCP reservation set for the Welltech device as you do not want the IP
changing on your gateway.
 	* The next page will be the port creation page, you will need to
configure your inbound routes on this page. Click next
 	* Here you will be needed to create your outbound routes for the
Welltech device.
 	* Finally proceed to the final page, go ahead and generate the
configuration file. This will need to be either saved or moved into
the directory below before your device will be able to sucessfully
provision.

C:ProgramData3CXInstance1DataHttpInterfaceprovisioningPROVSUBDIR 

STEP 3: PROVISIONING THE WELLTECHWELLGATE 2540
[HTTP://WWW.888VOIP.COM/PRODUCTS/WELLTECH-WELLGATE-2540.HTML] 

 	* Return to the web interface and log back in using the credentials
outlined earlier in this article.
 	* To provision this device we will be using the WAN port rather then
the LAN port and will have to get this configured to work correctly.

	* Proceed to the the Device Setting --> Network tab. You are going to
change the Network Type to DHCP and DNS Server Mode to Auto. This will
be pictured below. Go ahead and apply the changes. 

 	* [http://www.888voip.com/wp-content/uploads/2016/05/2540_1.png]

 	* Reboot the gateway and switch the gateway to the network it will
be located on, ensure that you have plugged into the WAN port.
 	* If you have not setup a MAC reservation for this device you will
have to locate the IP that it is on using a program such as Advanced
IP Scanner. Proceed to this IP and login to the web interface using
the details from above.

	* Proceed to the Device Setting --> Provisioning tab and configure as
follows: 

 	* Provisioning Type --> HTTP
 	* HTTP Config URL -->
http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR
[http://IPAddressofPBX:HTTPPortofPBX/provisioning/PROVSUBDIR]
 	* Refresh Interval (minute) --> 14400
 	* [http://www.888voip.com/wp-content/uploads/2016/05/2540_21.jpg]

 	* Apply these settings and proceed to the Maintenance tab, choose
reboot and apply your changes.
 	* On the reboot the device will download it's configuration from the
3CX server and your device will be able to make and receive calls when
complete. The ports will appear Green in the ports/trunks status tab
if the device took its configuration correctly.

 ]]></contentstext>
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<subject><![CDATA[Configuring Welltech Gateways with 3CX]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong></p>
<p>In this article, we will discuss how to recover a <a href="http://support.888voip.com/products/yealink-w52h.html">Yealink W52 handset</a> that is not functioning properly or will not turn on even after changing the batteries. Yealink has designed a tool specifically for upgrading and flashing the firmware on W52 handsets by using a USB cable. Prior to using this software, you will need to purchase a Mini-USB to USB 2.0 adapter. </p>
<p><strong><strong>Preparing your Computer: </strong></strong>Prior to upgrading your W52 Handset, <a href="http://support.yealink.com/documentFront/forwardToDocumentDetailPage?documentId=25">download the latest firmware version from Yealink</a>. The firmware you will need to download for the handset will begin with 26.*. You will also need to download the Yealink W52x USB Recovery Tool from our <a href="https://drive.google.com/folderview?id=0By_uJpOr3h2YTE55SHFwWlIzR0k&amp;usp=drive_web" target="_blank">Google Drive</a>. </p>
<p><strong>Preparing your W52 Handset: </strong>On the back of your W52 Handset, you will need to remove the battery cover, verify that your batteries are fully charged and then plug in the Mini-USB to USB 2.0 adapter as seen below:</p>
<p style="text-align: center;"><img src="http://support.888voip.com/wp-content/uploads/2016/05/rsz_w52.jpg" width="750" height="563" /></p>
<p> </p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/1.png" alt="" width="550" height="256" /></p>
<p> </p>
<ol>
<li>Plug the Mini-USB into your W52x handset as shown above and then plug the USB 2.0 side into your computer</li>
<li>Open the W52x Handset Firmware Update Tool and install the drivers
<ul>
<li><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/2-InstallDriver.png" alt="" width="300" height="164" /></li>
</ul>
</li>
<li>Restart your Computer after the drivers were successfully installed and disconnect the W52 handset from the USB port.
<ul>
<li><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/3-Restart.png" alt="" width="300" height="167" /></li>
</ul>
</li>
<li>After your computer reboots, open the W52 Handset Firmware Update Tool and plug in your W52 handset into your USB port again.</li>
<li>Your handset will now display in the application (Note: the Device Identification listed in the Device drop down may be all 0’s)<br />
<ul>
<li><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/7-Mac0-e1461697336985.png" alt="" width="550" height="234" /></li>
</ul>
</li>
<li><a href="http://support.yealink.com/documentFront/forwardToDocumentDetailPage?documentId=25">Download the latest W52 firmware from Yealinks website</a> and extract the ROM file into a folder ( The W52H firmware will begin with 26)</li>
<li><strong>Please verify that your W52 Handset is fully charged prior to proceeding to the next step or the upgrade may fail </strong></li>
<li>Press Upgrade and you will see a progress bar within the application; your device will reboot after it is done upgrading.<br />
<ul>
<li><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/5-Upgrade-e1461697384929.png" alt="" width="550" height="248" /></li>
</ul>
</li>
<li>After the upgrade is completed, your W52 will automatically restart and you will be able to verify that the firmware was successfully installed; navigate to the Menu on your Yealink W52H (press ok) and select Status -&gt; Handset. Confirm that the Handset Firmware is the same version that you installed using the update tool.</li>
</ol>
<p>If your device has already been upgraded to the same firmware you chose, you will receive the following message: Please verify that you have the correct ROM file selected and try upgrading again.</p>
<p><img style="display: block; margin-left: auto; margin-right: auto;" src="http://support.888voip.com/wp-content/uploads/2016/04/8-Equal.png" alt="" width="237" height="150" /></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: 

In this article, we will discuss how to recover a Yealink W52 handset
[http://support.888voip.com/products/yealink-w52h.html] that is not
functioning properly or will not turn on even after changing the
batteries. Yealink has designed a tool specifically for upgrading and
flashing the firmware on W52 handsets by using a USB cable. Prior to
using this software, you will need to purchase a Mini-USB to USB 2.0
adapter.  

PREPARING YOUR COMPUTER: Prior to upgrading your W52 Handset,
download the latest firmware version from Yealink
[http://support.yealink.com/documentFront/forwardToDocumentDetailPage?documentId=25].
The firmware you will need to download for the handset will begin with
26.*. You will also need to download the Yealink W52x USB Recovery
Tool from our Google Drive
[https://drive.google.com/folderview?id=0By_uJpOr3h2YTE55SHFwWlIzR0k&usp=drive_web]. 


PREPARING YOUR W52 HANDSET: On the back of your W52 Handset, you will
need to remove the battery cover, verify that your batteries are fully
charged and then plug in the Mini-USB to USB 2.0 adapter as seen
below: 

  

  

 	* Plug the Mini-USB into your W52x handset as shown above and then
plug the USB 2.0 side into your computer

	* Open the W52x Handset Firmware Update Tool and install the drivers 

 	* 

	* Restart your Computer after the drivers were successfully installed
and disconnect the W52 handset from the USB port. 

 	* 

 	* After your computer reboots, open the W52 Handset Firmware Update
Tool and plug in your W52 handset into your USB port again.

	* Your handset will now display in the application (Note: the Device
Identification listed in the Device drop down may be all 0’s)

 	* 

 	* Download the latest W52 firmware from Yealinks website
[http://support.yealink.com/documentFront/forwardToDocumentDetailPage?documentId=25]
and extract the ROM file into a folder ( The W52H firmware will begin
with 26)
 	* PLEASE VERIFY THAT YOUR W52 HANDSET IS FULLY CHARGED PRIOR TO
PROCEEDING TO THE NEXT STEP OR THE UPGRADE MAY FAIL 

	* Press Upgrade and you will see a progress bar within the
application; your device will reboot after it is done upgrading.

 	* 

 	* After the upgrade is completed, your W52 will automatically
restart and you will be able to verify that the firmware was
successfully installed; navigate to the Menu on your Yealink W52H
(press ok) and select Status -> Handset. Confirm that the Handset
Firmware is the same version that you installed using the update tool.

If your device has already been upgraded to the same firmware you
chose, you will receive the following message: Please verify that you
have the correct ROM file selected and try upgrading again. 

]]></contentstext>
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<subject><![CDATA[Flashing Firmware On A Yealink W52x Handset]]></subject>
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<contents><![CDATA[<p><strong>Overview:</strong> Network captures are one of the most useful tools while troubleshooting VoIP systems and resolving network related issues. Since more and more servers are being installed on Linux due to the minimal resource requirements, it's important to know how to obtain a Network Capture on a Linux server.</p>
<p><strong>Before you begin:</strong> You will need to have ROOT access to a Linux Server that is running the SSH (Secured SHell) Service. We will be using PuTTY to run commands on the server, and WinSCP to extract files from the server.</p>
<ol>
<li>First, let's install tcpump on your Linux server. Login as ROOT (Super User) and use the command:
<ul>
<li><em>sudo apt-get update &amp;&amp; apt-get install tcpdump</em></li>
<li>This command will download package lists and updates the list to get information on the newest versions of packages. After the list of packages have been updated, the command will proceed to download and install the package <i>tcpdump</i></li>
<li><em><img src="http://www.888voip.com/wp-content/uploads/2017/01/install.png" alt="" width="562" height="193" style="display: block; margin-left: auto; margin-right: auto;" /></em></li>
</ul>
</li>
<li>Ensure that no errors have been received. If you receive an error this may be due to your account permissions or you may be missing dependencies which need to be installed prior to installing tcpump.
<ul>
<li><img src="http://www.888voip.com/wp-content/uploads/2017/01/noerrors.png" alt="" width="586" height="191" style="display: block; margin-left: auto; margin-right: auto;" /></li>
</ul>
</li>
<li>You will want to change your directory to the area you would like to store files. Below are some of the commands you may use to create directories or navigate through your system.
<ul>
<li><em>pwd</em> command is used to Print Working Directory and will output the current directory you are in.</li>
<li><em>cd</em> command will be used to Change Directory.</li>
<li><em>mkdir</em> command is used to Make a Directory in the current working directory.</li>
<li><em>mv</em> command is used to move files from one location to another.</li>
</ul>
</li>
<li>Use the command below to start monitoring everything on your network (all ports and activity) and output to a file named "MyPCAP.pcap"
<ul>
<li><em>tcpdump -i any -w MyPCAP.pcap</em></li>
<li><em><img src="http://www.888voip.com/wp-content/uploads/2017/01/dump_start.png" alt="" width="646" height="68" style="display: block; margin-left: auto; margin-right: auto;" /><br /><br /></em></li>
<li>Use <em>Ctrl + C</em> to break the capture and return to the terminal.</li>
<li><img src="http://www.888voip.com/wp-content/uploads/2017/01/dump_end.png" width="642" height="117" style="display: block; margin-left: auto; margin-right: auto;" /></li>
</ul>
</li>
<li>Once your capture has ended, open WinSCP and connect to your server to copy your PCAP files to a different computer.</li>
<li>With WinSCP connected, navigate to the directory which you were working. See the above commands to assist with this.
<ul>
<li>Right click your file and click "Download" - this will download a copy to your local computer and keep the file on your Linux Server.</li>
<li><img src="http://www.888voip.com/wp-content/uploads/2017/01/download.png" alt="" width="474" height="402" style="display: block; margin-left: auto; margin-right: auto;" /></li>
</ul>
</li>
<li>Once the file is downloaded, you can use Wireshark to open the packet capture and troubleshoot network related issues.</li>
</ol>]]></contents>
<contentstext><![CDATA[OVERVIEW: Network captures are one of the most useful tools while
troubleshooting VoIP systems and resolving network related issues.
Since more and more servers are being installed on Linux due to the
minimal resource requirements, it's important to know how to obtain a
Network Capture on a Linux server. 

BEFORE YOU BEGIN: You will need to have ROOT access to a Linux Server
that is running the SSH (Secured SHell) Service. We will be using
PuTTY to run commands on the server, and WinSCP to extract files from
the server. 

	* First, let's install tcpump on your Linux server. Login as ROOT
(Super User) and use the command: 

 	* _sudo apt-get update && apt-get install tcpdump_
 	* This command will download package lists and updates the list to
get information on the newest versions of packages. After the list of
packages have been updated, the command will proceed to download and
install the package _tcpdump_
 	* __

	* Ensure that no errors have been received. If you receive an error
this may be due to your account permissions or you may be missing
dependencies which need to be installed prior to installing tcpump. 

 	* 

	* You will want to change your directory to the area you would like
to store files. Below are some of the commands you may use to create
directories or navigate through your system. 

 	* _pwd_ command is used to Print Working Directory and will output
the current directory you are in.
 	* _cd_ command will be used to Change Directory.
 	* _mkdir_ command is used to Make a Directory in the current working
directory.
 	* _mv_ command is used to move files from one location to another.

	* Use the command below to start monitoring everything on your
network (all ports and activity) and output to a file named
"MyPCAP.pcap" 

 	* _tcpdump -i any -w MyPCAP.pcap_

	* 

 	* Use _Ctrl + C_ to break the capture and return to the terminal.
 	* 

 	* Once your capture has ended, open WinSCP and connect to your
server to copy your PCAP files to a different computer.

	* With WinSCP connected, navigate to the directory which you were
working. See the above commands to assist with this. 

 	* Right click your file and click "Download" - this will download a
copy to your local computer and keep the file on your Linux Server.
 	* 

 	* Once the file is downloaded, you can use Wireshark to open the
packet capture and troubleshoot network related issues.

]]></contentstext>
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<subject><![CDATA[How to Get PCAPS from Linux]]></subject>
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<contents><![CDATA[<p>First, we need to configure the 3rd party VPN in Meraki. This needs to be done on both Meraki appliances. In the Meraki portal, select the proper network, then navigate to <strong>Security Appliance &gt; Site-to-site VPN</strong>.</p>
<p><img src="http://www.888voip.com/wp-content/uploads/2017/07/vpn-1.png" alt="" width="590" height="245" /><br /> <br /> From there, make sure the Type is set to <strong>Hub</strong> and the local subnets you supplied us earlier are set to <strong>Yes</strong>.</p>
<p><img src="http://www.888voip.com/wp-content/uploads/2017/08/vpn_settings.png" alt="" width="752" height="600" /><br /> <br /> From there, scroll down until you see Organization-wide settings. Next to the Non-Meraki VPN peers section, fill it out as follows. For the name, you can give it whatever name you like, it does not matter. Public IP will be the IP address we supply for the VPN. Private Subnets will be the subnet of the VPN, which we will also supply, along with the Preshares secret. For availability, this is not required. If your appliances already have a tag, you can use it, otherwise, you can leave this blank.<br /><br /><img src="http://www.888voip.com/wp-content/uploads/2017/07/vpn_3.png" alt="" width="1308" height="178" /> </p>
<p>Then click on the link underneath the IPsec policies. If you haven't yet edited the IPsec policies, it should say <strong>Default</strong>. Otherwise, it should show <strong>Custom</strong> as it does in the screenshot above.</p>
<p>In the new window that opens up, configure it as follows<br /><br /><img src="http://www.888voip.com/wp-content/uploads/2017/07/vpn-4.png" alt="" width="592" height="529" /></p>
<p>All of the above settings MUST match exactly, or the VPN tunnel will fail.</p>
<p>------------------------------</p>
<p>Once the above steps have been completed, please let us know so we can restart the tunnel connection and test to make sure the tunnel is functioning properly. Once we are on the same page, we will need to look at the Meraki Event Log on both Meraki appliances, to make sure that the tunnels have been negotiated successfully.</p>
<p>To bring up the event log in Meraki, navigate to <strong>Network-wide &gt; Event log</strong>.</p>
<p> <img src="http://www.888voip.com/wp-content/uploads/2017/07/vpn-5.png" alt="" width="594" height="231" /></p>
<p>Once we're in the event log, we're looking for a successful connection message. It should look something like this:<br /><br /><img src="http://www.888voip.com/wp-content/uploads/2017/07/vpn_6.png" alt="" width="1319" height="40" /></p>
<p>This is what we want to see. If we are getting other messages under the Non-Meraki / Client VPN negotiation filter, we will need to go back and do some tweaking</p>]]></contents>
<contentstext><![CDATA[First, we need to configure the 3rd party VPN in Meraki. This needs to
be done on both Meraki appliances. In the Meraki portal, select the
proper network, then navigate to SECURITY APPLIANCE > SITE-TO-SITE
VPN. 

 From there, make sure the Type is set to HUB and the local subnets
you supplied us earlier are set to YES. 

 From there, scroll down until you see Organization-wide settings.
Next to the Non-Meraki VPN peers section, fill it out as follows. For
the name, you can give it whatever name you like, it does not matter.
Public IP will be the IP address we supply for the VPN. Private
Subnets will be the subnet of the VPN, which we will also supply,
along with the Preshares secret. For availability, this is not
required. If your appliances already have a tag, you can use it,
otherwise, you can leave this blank.

  

Then click on the link underneath the IPsec policies. If you haven't
yet edited the IPsec policies, it should say DEFAULT. Otherwise, it
should show CUSTOM as it does in the screenshot above. 

In the new window that opens up, configure it as follows

All of the above settings MUST match exactly, or the VPN tunnel will
fail. 

------------------------------ 

Once the above steps have been completed, please let us know so we can
restart the tunnel connection and test to make sure the tunnel is
functioning properly. Once we are on the same page, we will need to
look at the Meraki Event Log on both Meraki appliances, to make sure
that the tunnels have been negotiated successfully. 

To bring up the event log in Meraki, navigate to NETWORK-WIDE > EVENT
LOG. 

  

Once we're in the event log, we're looking for a successful connection
message. It should look something like this:

This is what we want to see. If we are getting other messages under
the Non-Meraki / Client VPN negotiation filter, we will need to go
back and do some tweaking]]></contentstext>
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<subject><![CDATA[Setting up IPSEC Site-to-Site VPN on Cisco Meraki Security Appliance]]></subject>
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<contents><![CDATA[<p><strong>Overview</strong>: In this article we will go over the steps to upgrade your Free PBX Edition key to a Paid Standard, Professional or Enterprise Key.</p>
<p></p>
<p><strong>Methods</strong>: There are two different ways of upgrading you Free PBX edition Key <span>to a Paid Standard, Professional or Enterprise Key.</span></p>
<p></p>
<p>1) The first Method, which is the easiest way to upgrade your key is to simply have your sales rep/team upgrade your existing free key. Once the Key is upgraded, you simply need to reboot the server and your done</p>
<p></p>
<p>2) The second Method, if you already have a key you want to use or have purchased one; Requires an uninstall/reinstall of the PBX. The Most important part of this method, is how you take your backup.</p>
<p></p>
<p><strong>Changing your 3CX key to a New Key:</strong></p>
<p><strong></strong></p>
<p>1) When you Take the Backup of the server, you <strong>*must* </strong> uncheck the option to "Include license key information and FQDN" as shown below.</p>
<p></p>
<p><img src="http://www.888voip.com/wp-content/uploads/2017/10/backup.png" alt="Backup" width="957" height="338" /></p>
<p>2) Once the Backup has been completed and you have downloaded the backup to a safe location, you can simply uninstall the 3CX Server</p>
<p>3) Now you simply need to install 3CX and Upload your Backup file. Once your Backup file is loaded and you click next, you will be asked to enter your new 3CX key and continue your install as normal.</p>
<p></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: In this article we will go over the steps to upgrade your
Free PBX Edition key to a Paid Standard, Professional or Enterprise
Key. 

METHODS: There are two different ways of upgrading you Free PBX
edition Key to a Paid Standard, Professional or Enterprise Key. 

1) The first Method, which is the easiest way to upgrade your key is
to simply have your sales rep/team upgrade your existing free key.
Once the Key is upgraded, you simply need to reboot the server and
your done 

2) The second Method, if you already have a key you want to use or
have purchased one; Requires an uninstall/reinstall of the PBX. The
Most important part of this method, is how you take your backup. 

CHANGING YOUR 3CX KEY TO A NEW KEY: 

1) When you Take the Backup of the server, you *MUST*  uncheck the
option to "Include license key information and FQDN" as shown below. 

2) Once the Backup has been completed and you have downloaded the
backup to a safe location, you can simply uninstall the 3CX Server 

3) Now you simply need to install 3CX and Upload your Backup file.
Once your Backup file is loaded and you click next, you will be asked
to enter your new 3CX key and continue your install as normal. 

]]></contentstext>
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<subject><![CDATA[Updating your 3CX PBX Edition key to a Standard, Pro or Enterprise Key.]]></subject>
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<contents><![CDATA[<p><strong></strong><img src="http://www.888voip.com/wp-content/uploads/2011/04/sennheiser-featured-1.png" alt="" width="600" height="324" /></p>
<p><strong>Overview:</strong> This guide will provide you with the information and resources needed to upgrade the firmware on your <a href="https://www.888voip.com/products/sennheiser">Sennheiser </a>Bluetooth Headset and Dongle</p>
<p></p>
<p><strong>Let's Begin:</strong></p>
<p><strong>1)</strong> You will need to download and install the updater application on a windows machine, which can be <a href="http://sennheiser.com/headset-software-pc">downloaded Here</a> – Please make sure you download the end-user Version</p>
<p></p>
<p><strong>2)</strong> Once you have installed the program on your computer, you will then need to connect the device via USB to the computer and run the program. To do so, remove the ear pad on the same side of the headset as the microphone boom arm to reveal the micro-USB port located underneath and then use a micro-USB to USB cable to connect it with your PC.</p>
<p></p>
<p><img src="http://www.888voip.com/wp-content/uploads/2018/01/sennheiser-2.jpg" alt="" width="600" height="450" /></p>
<p><br /> <strong>3)</strong> When you launch the program, you will see the headset/dongle device connected to your PC. It will display both the current firmware version on your device as well as the latest version available. (If you are having trouble seeing your headset, please try another micro USB cable and/or factory reset your device and power it back on, <a href="https://sennheiser.zendesk.com/hc/en-us/articles/218556097-How-to-perform-a-factory-reset-with-MB-Pro-series">following this guide</a>) Now simply check the box that corresponds with your headset/dongle and click the 'Update Selected' button and the program will download the latest version and upload it to the headset/dongle.</p>
<p></p>
<p><strong></strong></p>
<p><img src="http://www.888voip.com/wp-content/uploads/2018/01/sennheiser-1.png" alt="" width="600" height="338" /></p>
<p></p>
<p><strong>4)</strong> Congratulations, you have completed the update of your headset and dongle; you will now see your Current Version and new versions match. <strong> </strong></p>
<p><strong></strong></p>
<p><strong><img src="http://www.888voip.com/wp-content/uploads/2018/01/sennheiser-3.png" alt="" width="600" height="336" /></strong></p>]]></contents>
<contentstext><![CDATA[OVERVIEW: This guide will provide you with the information and
resources needed to upgrade the firmware on your Sennheiser
[https://www.888voip.com/products/sennheiser]Bluetooth Headset and
Dongle 

LET'S BEGIN: 

1) You will need to download and install the updater application on a
windows machine, which can be downloaded Here
[http://sennheiser.com/headset-software-pc] – Please make sure you
download the end-user Version 

2) Once you have installed the program on your computer, you will then
need to connect the device via USB to the computer and run the
program. To do so, remove the ear pad on the same side of the headset
as the microphone boom arm to reveal the micro-USB port located
underneath and then use a micro-USB to USB cable to connect it with
your PC. 

 3) When you launch the program, you will see the headset/dongle
device connected to your PC. It will display both the current firmware
version on your device as well as the latest version available. (If
you are having trouble seeing your headset, please try another micro
USB cable and/or factory reset your device and power it back on,
following this guide
[https://sennheiser.zendesk.com/hc/en-us/articles/218556097-How-to-perform-a-factory-reset-with-MB-Pro-series])
Now simply check the box that corresponds with your headset/dongle and
click the 'Update Selected' button and the program will download the
latest version and upload it to the headset/dongle. 

4) Congratulations, you have completed the update of your headset and
dongle; you will now see your Current Version and new versions
match.   

]]></contentstext>
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<subject><![CDATA[Upgrading The firmware on your Sennheiser Bluetooth Headset]]></subject>
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