AXIS C6110 Network Paging Console

The following video shows an example of how you can install an AXIS C6110 Network Paging Console together with an AXIS TC6901 Gooseneck Microphone.

For complete instructions on all installation scenarios as well as important safety information, see the installation guide on axis.com/products/axis-c6110/support.

To find Axis devices on the network and assign them IP addresses in Windows®, use AXIS IP Utility or AXIS Device Manager. Both applications are free and can be downloaded from axis.com/support.

For more information about how to find and assign IP addresses, go to How to assign an IP address and access your device.

You can use the device with the following browsers:

✓: Recommended
*: Supported with limitations

1. Open a browser and type the IP address or host name of the Axis device.

   If you don’t know the IP address, use AXIS IP Utility or AXIS Device Manager to find the device on the network.

2. Type the username and password. If you access the device for the first time, you must create an administrator account. See Create an administrator account.

For descriptions of all features and settings in the web interface of devices with AXIS OS, see AXIS OS web interface help.

The first time you log in to your device, you must create an administrator account.

1. Enter a username.

2. Enter a password. See Secure passwords.

3. Re-enter the password.

4. Accept the license agreement.

5. Click Add account.

The device password is the primary protection for your data and services. Axis devices do not impose a password policy as they may be used in various types of installations.

To protect your data we strongly recommend that you:

• Use a password with at least 8 characters, preferably created by a password generator.

• Don’t expose the password.

• Change the password at a recurring interval, at least once a year.

1. To make sure that the device has its original AXIS OS, or to take full control of the device after a security attack:

2. Reset to factory default settings. See Reset to factory default settings.

   After the reset, secure boot guarantees the state of the device.

3. Configure and install the device.

Use peer-to-peer when the communication is between a few user agents within the same IP network and there is no need for extra features that a PBX-server could provide. To better understand how P2P works, see Peer-to-peer SIP (P2PSIP).

For more information about setting options, see .

1. Go to System > SIP > SIP settings and select Enable SIP.

2. To allow the device to receive incoming calls, select Allow incoming calls.

3. Under Call handling, set the timeout and duration for the call.

4. Under Ports, enter the port numbers.

5. • SIP port – The network port used for SIP communication. The signaling traffic through this port is non-encrypted. The default port number is 5060. Enter a different port number if required.

   • TLS port – The network port used for encrypted SIP communication. The signaling traffic through this port is encrypted with Transport Layer Security (TLS). The default port number is 5061. Enter a different port number if required.

   • RTP start port – Enter the port used for the first RTP media stream in a SIP call. The default start port for media transport is 4000. Some firewalls might block RTP traffic on certain port numbers. A port number must be between 1024 and 65535.

6. Under NAT traversal, select the protocols you want to enable for NAT traversal.

7. Note

   Use NAT traversal when the device is connected to the network from behind a NAT router or a firewall. For more information see NAT traversal.

8. Under Audio, select at least one audio codec with the desired audio quality for SIP calls. Drag-and-drop to change the priority.

9. Under Additional, select additional options.

10. • UDP-to-TCP switching –  Select to allow calls to switch transport protocols from UDP (User Datagram Protocol) to TCP (Transmission Control Protocol) temporarily. The reason for switching is to avoid fragmentation, and the switch can take place if a request is within 200 bytes of the maximum transmission unit (MTU) or larger than 1300 bytes.

    • Allow via rewrite – Select to send the local IP address instead of the router's public IP address.

    • Allow contact rewrite – Select to send the local IP address instead of the router's public IP address.

    • Register with server every – Set how often you want the device to register with the SIP server for the existing SIP accounts.

    • DTMF payload type – Changes the default payload type for DTMF.

11. Click Save.

Use a PBX-server when user agents will communicate within and outside the IP network. Additional features could be added to the setup depending on the PBX-provider. To better understand how P2P works, see Private Branch Exchange (PBX).

For more information about setting options, see .

1. Request the following information from your PBX provider:

2. • User ID

   • Domain

   • Password

   • Authentication ID

   • Caller ID

   • Registrar

   • RTP start port

3. To add a new account, go to System > SIP > SIP accounts and click + Account.

4. Enter the details you received from your PBX provider.

5. Select Registered.

6. Select a transport mode.

7. Click Save.

8. Set up the SIP settings the same way as for peer-to-peer. See Set up direct SIP (P2P) for more information.

To add contacts, open the web interface by entering the IP address of your paging console in a web browser.

1. Add an individual device as a recipient:

2. Go to Communication > Recipients > Devices.

3. Click Add device.

4. Enter the details and click Save.

   For information about the options under Protocol, see .

1. Add an individual person as a recipient:

2. Go to Communication > Recipients > Contacts.

3. Click Add contact.

4. Enter the details and click Save.

   For information about the options under Protocol, see .

1. Create a group of VAPIX recipients:

2. Go to Communication > Recipients > Groups.

3. Click Add group.

4. Enter the details and click Save.

To configure buttons and folders, open the web interface by entering the IP address of your paging console in a web browser.

1. Create a new button or folder:

2. Go to the location where you want to add the button or folder.

   This will either be on the Home view or inside one of your folders.

3. Click a white button.

   White color indicates that the button has not been configured.

4. Select if you want to create an action or a folder.

5. Note

   If you have a view that is located deep down in the folder structure, a good practice is to add a Home button that makes it easy to return to the home view.

6. Enter the details and click Save.

Edit or delete an existing button or folder:

• Click and select Edit or Delete.

Rename the home view title:

1. Click next to the home view title.

2. Select Rename title.

3. Enter the new title and click Save.

Add a new page:

• Click Add page.

  This will add a page to the same location, i.e. at the Home view or inside the current folder.

1. Create a VAPIX recipient:

2. 1. Go to Communication > Recipients.

   2. If you want to add a device, go to Devices.

      If you want to add a contact, go to Contacts.

   3. Click + Add device or + Add contact.

   4. Name your recipient.

   5. Under Protocol, select VAPIX.

   6. Enter the IP address for the recipient.

   7. Enter the user name and password for the recipient.

   8. Click Save.

3. Create a two-way action:

4. 1. Go to Display > Configuration > Actions.

   2. Click + Add action.

   3. Under Action, select Two-way.

   4. Under Contact, select your VAPIX recipient.

   5. Click Save.

5. Configure a button:

6. 1. Go to Display > Configuration > Buttons.

   2. Click an available button.

   3. Under Select button type, select Action.

   4. Under Select an action to be triggered by the button, select Use an existing action.

   5. Click the row of your two-way action in the list.

   6. Click Save.

When you press the configured button on your AXIS C6110 Paging Console, a two-way VAPIX call will be made to the recipient.

Remember that the microphone on the recipient device must be activated. Enable echo cancellation on the recipient device to enhance the quality of the two way call. See Device settings.

You can use AAM Edge to configure a button on the C6110 to page one or several physical zones.

1. Open AXIS Audio Manager Edge.

2. Create a paging recipient.

3. Open the web interface

4. set it to be one-way

5. assign the desired zone/zones

6. Open the recipient to see which Intermediary device has been chosen

7. Copy the IP adress of the intermediary device.

8. Return to the web interface of C6110.

9. Go to Communication > Recipients > Devices and click + Add device

10. Give the Contact a name, select SIP as a protocol, enter the IP adress in the SIP Address field, and select the peer-to-peer account on the C6110.

11. Go to Display -> Configuration and add a new button.

12. Create new action -> Action: One way, Contact: The Contact created in the step above. Save the button.

To change the display settings, open the web interface by entering the IP address of your paging console in a web browser.

• To adjust brightness, timers and presence detection, go to Display settings > Display.

• To adjust language and clock settings for the display of your paging console, go to Display > Localization.

For more information about the individual options, see .

You can create rules to make your device perform actions when certain events occur. A rule consists of conditions and actions. The conditions can be used to trigger the actions. For example, the device can play an audio clip according to a schedule or when it receives a call, or send an email if the device changes IP address.

To learn more, see Get started with rules for events.

1. Navigate to the page on the display where the contact is located.

   Contacts are indicated by .

2. To place a call, press the button for the contact.

3. To mute or unmute your microphone, press the Mute or Unmute button.

4. To regulate the volume level of your speaker, press the volume button on the left side of the paging console.

5. To end the call, press the button for Hang up.

When you receive a call, the display shows Incoming call and a ringing signal is heard.

1. To answer the call, press the Answer button.

2. To hang up or reject the call, press the Hang up button.

If you have missed a call, is shown at the top right corner of the display. To see who called, press the Call history button.

To page a one-way live callout:

1. Navigate to the page on the display where the target is located.

   The target can be an individual person or device, or a group. Targets are indicated by .

2. Press the button of the target.

3. Wait for the pre-announcement message to be played, if such message is configured for the target.

4. Press and hold the push-to-talk button, and speak your message.

5. When you are done, press Cancel.

Play a pre-recorded audio file:

1. Navigate to the page on the display where the announcement is located.

   Announcements are indicated by .

2. Press the button for the announcement.

The AXIS TC6901 Gooseneck Microphone is an accessory that is sold separately.

For mounting instructions, see the installation guide for AXIS TC6901 Gooseneck Microphone.

To use a gooseneck microphone:

1. Open the web interface by entering the IP address of your paging console in a web browser.

2. Go to Device settings.

3. Set Input type to Balanced microphone.

You can connect a headset to the 3.5 mm audio connector located on the side of the AXIS C6110 Network Paging Console.

You can adjust the volume for the headset using the volume buttons.

If you connect headphones without a microphone, the internal microphone will stay active.

The Session Initiation Protocol (SIP) is used to set up, maintain and terminate VoIP calls. You can make calls between two or more parties, called SIP user agents. To make a SIP call you can use, for example, SIP phones, softphones or SIP-enabled Axis devices.

The actual audio or video is exchanged between the SIP user agents with a transport protocol, for example RTP (Real-Time Transport Protocol).

You can make calls on local networks using a peer-to-peer setup, or across networks using a PBX.

The most basic type of SIP communication takes place directly between two or more SIP user agents. This is called peer-to-peer SIP (P2PSIP). If it takes place on a local network, all that’s needed are the SIP addresses of the user agents. A typical SIP address in this case would be sip:<local-ip>.

Example

You can set up a SIP-enabled phone to call an audio device on the same network using a peer-to-peer SIP setup.

When you make SIP calls outside your local IP network, a Private Branch Exchange (PBX) can act as a central hub. The main component of a PBX is a SIP server, which is also referred to as a SIP proxy or a registrar. A PBX works like a traditional switchboard, showing the client's current status and allowing for example call transfers, voicemail, and redirections.

The PBX SIP server can be set up as a local entity or offsite. It can be hosted on an intranet or by a third party provider. When you make SIP calls between networks, calls are routed through a set of PBXs, that query the location of the SIP address to be reached.

Each SIP user agent registers with the PBX, and can then reach the others by dialing the correct extension. A typical SIP address in this case would be sip:<user>@<domain> or sip:<user>@<registrar-ip>. The SIP address is independent of its IP address and the PBX makes the device accessible as long as it is registered to the PBX.

Example

Use NAT (Network Address Translation) traversal when the Axis device is located on an private network (LAN) and you want to access it from outside of that network.

Each NAT traversal protocol can be used separately or in different combinations depending on the network environment.

• ICE The ICE Interactive Connectivity Establishment) protocol increases the chances of finding the most efficient path to successful communication between peer devices. If you also enable STUN and TURN, you improve the ICE protocol’s chances.

• STUN - STUN (Session Traversal Utilities for NAT) is a client-server network protocol that lets the Axis device determine if it is located behind a NAT or firewall, and if so obtain the mapped public IP address and port number allocated for connections to remote hosts. Enter the STUN server address, for example, an IP address.

• TURN - TURN (Traversal Using Relays around NAT) is a protocol that lets a device behind a NAT router or firewall receive incoming data from other hosts over TCP or UDP. Enter TURN server address and the login information.

To read about all the features and settings available in the web interface of devices with AXIS OS, go to AXIS OS web interface help.

For SD card recommendations, see axis.com.

SD, SDHC, and SDXC Logos are trademarks of SD-3C LLC. SD, SDHC and SDXC are trademarks or registered trademarks of SD-3C, LLC in the United States, other countries or both.

• The control button is used for:

• Calibrating the speaker test. Press and release the control button and a test tone is played.

• Resetting the product to factory default settings. See Reset to factory default settings.

RJ45 Ethernet connector with Power over Ethernet (PoE).

3.5 mm input/output connector for headset (4-pole TRRS) or headphone (3-pole TRS).

Audio input/output for headset (standard)

For more information, see Use an AXIS TC6901 Gooseneck Microphone

Use the I/O connector with external devices in combination with, for example, motion detection, event triggering, and alarm notifications. In addition to the 0 VDC reference point and power (12 V DC output), the I/O connector provides the interface to:

Digital input

For connecting devices that can toggle between an open and closed circuit, for example PIR sensors, door/window contacts, and glass break detectors.

Digital output

For connecting external devices such as relays and LEDs. Connected devices can be activated by the VAPIX® Application Programming Interface, through an event or from the device’s web interface.

6-pin terminal block

To reset the product to the factory default settings:

1. Disconnect power from the product.

2. Press and hold the control button while reconnecting power. See Product overview.

3. Keep the control button pressed for 10 seconds until the status LED indicator turns amber for the second time.

4. Release the control button. The process is complete when the status LED indicator turns green. If no DHCP server is available on the network, the device IP address will default to one of the following:

5. • Devices with AXIS OS 12.0 and later: Obtained from the link-local address subnet (169.254.0.0/16)

   • Devices with AXIS OS 11.11 and earlier: 192.168.0.90/24

6. Use the installation and management software tools, assign an IP address, set the password, and access the product.

You can also reset parameters to factory default through the device’s web interface. Go to Maintenance > Factory default and click Default.

If you need more help, go to axis.com/support.