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Technical Hints

How to configure Polycom VVX410 with Plantronics CS520 and EHS APP-51

List of Components: 

  1. Power connector for Plantronics Base
  2. Plantronics Headset
  3. APP-51 EHS Adapter
  4. Plantronics Base

Configuration:

  1. Connect the power connector (1) to the Plantronics Base (4) via the back of the base.

 

2. Connect the EHS Adapter (3) into the back of the Plantronics Base (4).

3. Connect the other ends of the EHS Adapter (3) into the back of the Polycom VVX410. Make sure that you connect the cable to the headset port and not the handset port by mistake.

4. In order to make sure that the headset works correctly with the phone you need to go to :

Settings > Basic Settings > Preferences > Headset Mode > Select Plantronics EHS Mode.

5. Make sure that the Dial settings on the bottom of the Plantronics base are set correctly. The correct Dial settings are below:

The headset should now work with the Polycom phone.

How to Enable ZTP on Polycom Phones

Here is a step by step guide on how to enable ZTP on a range or polycom Phones:

  1. Plug the device in and wait for the screen to display “Loading Application” and then click cancel as soon as it appears.
  2. Wait for the screen to display “Welcome Auto Boot” and click on “Setup” and type in your password, by default this should be 456 unless it has been changed and then press “OK”
  3. Select “Provisioning Server” and then scroll down to the bottom to where it says “ZTP” to enable this setting by clicking select  and then use the down arrows to change the “ZTP” to “Enabled”
  4. Once this is done keep clicking “Exit” and then “Save & Reboot”, you have successfully enabled ZTP.

If you have any problems with this please contact our technical support team.

Jablocom Essence – Headset Compatibility

Did you know that you can use a headset with the Jablocom Essence phone? We have performed testing using an Eartec 308 monaural headset with a EAR-QD002P which is the compatible bottom cable for the headset with the Essence. Please see the following guide below to find out how to connect and use the headset:

Step 1:

You will need to disconnect the handset curly cable from the phone which is located at the back of the phone.

Step 2:

Connect the EAR-QD002P bottom cable into the RJ9 port where you originally disconnected the handset curly cable from.

Step 3:

You will need to ensure that the Essence is turned on and then navigate to the menu by pressing the key labelled ‘OPTIONS’ on the phone keypad.

Step 4:

Using the arrow keys, scroll down to ‘Settings’, then press the green button to proceed to the next menu.

Step 5:

You will then need to select ‘Phone’ which should be the first option in the menu, then scroll down to ‘Headset active’ option and press the green button.

When an incoming call is received on the phone, you can now press the green button to answer the call on the headset.

Sangoma PBXact UC – Enabling Phone Apps

Sangoma’s PBXact UC systems enable you to remotely manage deployed phones from your computer, but did you know there’s also a range of Apps available, designed to enhance the capabilities of Sangoma IP phones? We’ve put the following step-by-step guide together to enable you to discover how you can start utilising the great features available.

Please note that this guide shows you how to enable Phone Apps, it doesn’t show the process of creating an extension. If you would like to find out how to create an extension, please see our earlier blog post on how to create an extension using PBXact

Step 1:
You will need to start by web browsing to the IP address of your Sangoma PBXact UC system. If you don’t already know the IP address of the PBXact UC system, you may need to log into your router to view a list of active network devices.

Step 2:
Once you have browsed to the IP address of your PBXact UC system, you should see the following options:


You will need to go ahead and select the ‘PBX Admini
stration’ option. You should then be prompted to enter a username and password in order to login to the main interface of the system.

If you don’t know the username and password, you may need to consult an administrator or relevant person to help you with this.

Step 3:
Providing you have entered the correct username and password, you should now see the homepage showing the ‘System Overview’, ‘PBXact Statistics’, ‘Uptime’ and options at the top of the page.

You can either hover over ‘Modules’ located at the top of the screen in ‘Advanced Mode’ and select ‘User Management’, or use the magnifying glass on the right hand side and search for ‘User Management’ and select the option when it appears below the search field.

Step 4:
You should now see the ‘User Manager’ screen which may list users below if you already have users registered on the system.
Your screen should look like the screenshot to the right.

Step 5:
Please note that this step will involve creating a new user. You will need to click on the ‘Add’ button as shown in the screenshot above.


This should then take you to the screenshot shown above. You will then need to enter a ‘Login Name’, ‘Password’ and if you are linking this to an extension that already exists on the system, you will need to select the ‘Primary Linked Extension’ which should display a drop down list of extensions. Please note that extensions already assigned to other users will not be displayed in the drop down list.

Step 6:
Using the navigation bar under ‘Add User’, you will need to navigate to the ‘Phone Apps’ section, as can be seen below.

When you navigate to ‘Phone Apps’, you should see the ‘General’ section underneath and a section called ‘Allow Access’. You will need to change this option to ‘Yes’. This allows access to the Phone Apps for the user you are editing.

Step 7:
Depending on which Phone Apps you would like to set up for the user, you will need to navigate across the different tabs and enable the applications that you would like to assign to that particular user.

If you want to enable access to call queues, you will need to navigate to ‘Queues’ and change ‘Enable Queue Access’ to ‘Yes’. If you would also like to setup queue agent access, you will need to change ‘Enable Queue Agent Access’ to ‘Yes’. You will then also need to select the queues from a list under ‘Allowed Queues’. Please see the following screenshot below that shows this:


As you can see from the screenshot above, there is ‘TestQueue <6000>’ is shown in the list under ‘Allowed Queues’. Please note that you must have created a queue before hand in order for it to be shown in the list.

Step 8:
You may want to enable the login and logout feature for that user so that they can log out of the Sangoma IP phone once they have finished using it. In order to do this, you will need to navigate to ‘Other Apps’ and change ‘Enabled Login/Logout Access’ to ‘Yes’. Please see below a screenshot that shows this:

You may also want to allow Do Not Disturb (DND) depending if this is required or not. You can simply change the options that you would like to enable to ‘Yes’. Please note that additional configuration may be required when enabling certain Phone Apps.

Step 9:
Once you have finished configuring the user with Phone Apps, you can go ahead and submit your changes by pressing the ‘Submit’ button located near the bottom right hand side of the page.

You should then be taken back to the ‘User Manager’ section which shows you a list of users. You will then need to click ‘Apply Config’ which should be displayed near the top right hand side of the page. This writes the changes made to the PBXact UC system.

That’s the end of the configuration process.

Providing you have a valid extension and user set up, you should now be able to see the Phone Apps on the phone once provisioned/configured correctly.

SIP-TLS with the Panasonic TGP600

The Panasonic TGP-600 DECT phone supports encryption of SIP signalling and audio (RTP) using the common SIP-TLS and SRTP methods supported by many VoIP platforms.

Configuration is very simple.  In the SIP Settings page:

Important settings are:

  • Proxy Server Port, Registrar Server Port, Presence Server Port.  The standard port for encrypted SIP is 5061 (rather than 5060 for normal plain-text SIP).  This depends on your SIP platform.
  • Transport Protocol. Set this to TLS
  • TLS Mode.  Depends on your platform but SIP-TLS is what I am using with an Asterisk PBX

All other settings on that page are the as normal.  You might need to alter some of the SRTP settings for voice encryption, on the VoIP Settings page:

  • SRTP Mode. This also depends on your SIP platform but Asterisk doesn’t handle negotiation of encryption so if it is being used at all, you need to get the phone to always use it, not attempt to negotiate.  In that case, this setting is set to “SRTP”

Certificates

By default the Panasonic phone is set to accept all certificates (meaning that self-signed certificates will work OK).  You can provision the phones to verify the certificate if you want to using the setting SIP_TLS_VERIFY_1_=”1″.  You need to ensure that you have loaded the necessary root certificate beforehand.

Why use TLS & SRTP?

Security:  If you are able to sniff the traffic on someone’s network (e.g. using Wireshark or tcpdump) then you will capture any VoIP calls going on.  A tool such as Wireshark can be used to extract the audio from the RTP packets on the network.  The SIP packets can be read in plain-English and can be used to ascertain certain things such as what extension numbers there are, who is phoning different numbers etc…

If the SIP traffic is encrypted then no-one can see it other than the telephone and the SIP server at the other end (much like HTTPS used by secure websites).

If the RTP stream is encrypted then the audio cannot be extracted from the network without access to the SRTP keys generated on each call.  If you try this using Wireshark, the audio file you’ll get out of it will contain only white-noise.  Because the encryption keys for SRTP are generated on each call and send within the SIP packets, it would make no sense to use SRTP without encrypting the SIP packets as well.

Hiding SIP from Application Layer Gateways:  Routers with SIP-ALGs built into them are the biggest single cause of issues with SIP, things such as one-way audio, calls cutting out, calls failing to connect etc…. can all be caused by a SIP-ALG on a router.  The job of the ALG is to keep an eye out for SIP packets going past and then to modify them in an attempt to fix them up to work through NAT.  But they nearly always cause more problems than they solve.  A less obvious attraction to SIP-TLS is that if the SIP traffic is encrypted, then a SIP-ALG cannot possibly see any SIP traffic going through it and much less, make any modifications to it.  This can be very useful for remote phones talking to a hosted PBX or a central office PBX.

The latter advantage is the main reason I am seeing people interested in SIP-TLS or already using it, rather than it’s intended use which is for secure calling.

Configure Door Release Button on Grandstream GXV3275 to work with 2N Helios IP Force

Did you know that you can configure a programmable key on the Grandstream GXV3275 that will allow you to unlock the door via a peer-to-peer call from the 2N Helios IP Force to the Grandstream GXV3275 from the press of a button? Please see the following guide below in order to find out how to do this:

Step 1:
The first step is to start by web browsing to the IP address of the Grandstream GXV3275 using a web browser of your choice e.g. Google Chrome, Firefox.

Step 2:
Providing you have entered the correct IP address, you should see the following page prompting you to login to the web interface of the phone:

If you don’t know the username and password to login, you may need to consult an IT administrator.

Step 3:
Providing you have entered the correct username and password, you should see a few different settings to choose from. You will need to navigate to ‘Account’ using the top navigation bar, then ‘Call Settings’ using the navigation bar to the left hand side.

Step 4:
You will need to scroll down to the ‘Programmable Keys’ section which can be found near the bottom of the interface. You will need to set ‘Key Mode’ to ‘Dial DTMF’ using the drop down list.

Step 5:
For ‘Name’, you will need to enter a suitable name that will be displayed on the door release button on the phone.

Step 6:
For ‘DTMF Content’, you will need to enter the code that is set under ‘Switch Codes’ on the web interface of the 2N Helios IP Force. The default switch code that is set on the IP Force is ’00’.

Step 7:
Because the Grandstream GXV3275 and IP Force are peer-to-peer calling, you won’t need to worry about the ‘Dial DTMF Condition’. The default option should be set to ‘Incoming/Outgoing Call’. You can go ahead and leave that as the selected option.

Step 8:
Once you have configured all of the above options, you can go ahead and hit the ‘Save’ button at the bottom of the page. You should then be taken to the top of the page prompted to apply the configuration changes.

Firmware/System Version Used:

Grandstream GXV3275 Android Phone: System Version: 1.0.3.144

2N Helios IP Force: Firmware: 2.18.1.27.8

Yealink W52P & RTX 4002P – Setting up Daisy Chain

Yealink W52P & RTX 4002P – Setting up Daisy Chain

*RTX4022P can not be set up in Daisy Chain with 3rd party devices such as Gigaset and Yealink*

* Requires RTX PC Software *
* Maximum of 3 Repeaters in a Daisy Chain *

1) Enable Repeater mode on the W52P:

– Press OK on the W52H ‘handset’ and then go to settings. Choose “System Settings” and then select “Repeater Mode”.

2) Pair the first RTX repeater in the chain to the base:

– Plug in the RTX repeater for 1-5 seconds (We count to 3) and then unplug it.

– Plug it in again and leave it. The LED indicator on the RTX repeater will then flash slowly (This means the repeater is ready for manual registration).

– Press the paging key on the W52P base until the “Handset” LED light on the base starts flashing.

– The LED on the RTX repeater will now change to flash faster than it did before.

3) Assign a repeater number to the RTX repeater from 2 to 7, for up to 6 repeaters. (The W52P takes repeater number 1)

– Dial #*9 on the W52H ‘Handset’, it will stay on this call for the duration of this procedure. The LED on the repeater will be lit solid.

– Press a number between 2 and 7. The LED indicator on the repeater will flash a corresponding number of times and then stay solid.

– Accept the registration by pressing the * key and end the call on the W52H ‘Handset’.

4) Configure the repeater:

– Connect this repeater to the RTX USB cable, power supply and your PC.

– Load the RTX software on the supplied CD.

– Select the COM port the USB cable is connected to.

– Press the “load” button.

– Write down the “RFPI” number from the “Repeater” section.

– Disconnect this repeater but do not close the software.

4.1) Configure the next repeaters

– Connect the next repeater that will be used in the daisy chain.

– Type the “RFPI” number you have written down into the “RFPI” box in the “Network Device” section.

– Tick the “Ignore Hop Control” box and optionally the “Monitor Beep” if you want that on (useful for testing).

– Increment the “RPN” number in the “Repeater” section. This need to go up by one for each repeater in this chain.

– Press “Save”.

– If this isn’t the last repeater in the chain, click “Load” and write down the “RFPI” number from the “Repeater” section (This will then be used by the next repeater in the chain in it’s “Network Device” section).

– Pair this repeater to the base by following the procedure in steps 2) & 3).

* Repeat step 4.1) for each repeater in this chain.

* Repeat steps 2) to 4.1) for each chain.

How to make the 2N unit sound an alarm when the tamper switch is triggered

When the tamper switch is triggered you can set it up to alert sound from the 2N unit, this would be useful for security purposes. Here is a step by step guide on how to set this up.

If you have a 2N tamper switch and want to know how to install it into your Verso, please click this link:

2N’s Helios IP Verso Tamper Switch

After the installation of the tamper switch you will need to configure it. Go on the 2N web-interface and go to Services (Purple tile) > Automation, from this page you will need to tick the box “Function Enabled” and fill in the “Object Type” and “Parameters” as seen below:

For this alarm I have used a dog barking which = *3 but there are other sounds to choose from here:

2N-Action-PlayUserSounds

Once you have chosen the sound you want the unit to play when the Tamper switch is triggered, click “Apply” at the bottom and you have finished. See me test this automation below:

If you have any problems with this please contact our support team.

How to receive an E-mail from a 2N Verso when button is pressed.

You are out of the office and only have access to E-mails but you still want to be notified when someone is at the front door. Creating this simple automation will allow you to receive an E-mail with the time and date of when someone presses the button on the 2N Verso. You can also get it to send you a picture of who it is that pressed the button.

Go on the 2N web-interface and go to Services (Purple tile) > Automation, from this page you will need to tick the box “Function Enabled” and fill in the “Object Type” and “Parameters” as seen below:

Event=1; Sender=2nVerso@frontdoor.com; Email=YourEmailAdress; Subject=Someone has tried calling you; (Subject can be changed to say what you want)

Go to the “E-Mail” tab, this is were you will need to put your smtp server address along with your email address and password. If you are using your gmail account then the smtp server and port are most likely going to be as follows:

screenshot-from-2016-11-14-113436

For more information about the smtp server settings go to this page: https://wiki.2n.cz/hip/conf/latest/en/5-konfigurace-interkomu/5-4-sluzby/5-4-4-e-mail

You have now finished, you can now test this feature by pressing the intercom button and watch your inbox for the email to come through, as shown below: 

If you have any issue please contact our support team.

Cisco IP 8800 Key Expansion Module Key Programming

With the Cisco 8880 Expansion Module you are able to program keys to show names of other phone users.

The key settings can be found in the web interface of the phone on the “Unit1-3” tabs depending on how many extension modules you have.

 

The code to display a name on the extension module is “fnc=blf+cp;sub=5070@$PROXY;ext=5070@$PROXY;nme=Name”.

The “;nme=Name” is where you can type the text you want to appear on the extension module.

The “;sub=5070” and “;ext=5070” is where the extension number must go.