AXIS C2005 Network Ceiling Speaker

This manual describes how you make the device accessible to your audio system, and how to configure the device directly from its interface.

If you are using an audio or video management software, you can use that software for configuring the device. The following management software are available for controlling your audio system:

• AXIS Audio Manager Edge — Audio management software for small systems. Comes pre-installed on all audio devices with a firmware equal to or higher than 10.0.

• • AXIS Audio Manager Edge user manual

• AXIS Audio Manager Pro — Advanced audio management software for large systems.

• • AXIS Audio Manager Pro user manual

• AXIS Camera Station Pro — Advanced video management software for large systems.

• • AXIS Camera Station Pro user manual

• For more information, see Audio management software.

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 enter the IP address or host name of the Axis device.

2. Enter the username and password. If you access the device for the first time, you must set the root password. See Set a new password for the root account.

1. Type a password. Follow the instructions about secure passwords. See Secure passwords.

2. Retype the password to confirm the spelling.

3. Click Save. The password has now been configured.

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.

You can run a speaker test to verify from a remote location that a speaker is working as intended. The speaker performs the test by playing a series of test tones that are registered by the built-in microphone. Every time you run the test, the registered values are compared with the values that were registered during the calibration.

1. Go to the device interface > Audio > Speaker test.

2. To calibrate the audio device, click Calibrate.

3. Note

   Once the Axis product is calibrated, the speaker test can be run at any time.

4. To run the speaker test, click Run the test.

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.

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.

In this example the audio device is configured to send an email to a defined recipient when a speaker test fails. The speaker test is configured to be performed 18:00 every day.

1. Set up a schedule for the speaker test:

2. 1. Go to the device interface > System > Events > Schedules.

   2. Create a schedule that starts at 18:00 and ends at 18:01 every day. Name it "Daily at 6pm".

3. Create an email recipient:

4. 1. Go to the device interface > System > Events > Recipients.

   2. Click Add recipient.

   3. Name the recipient "Speaker test recipients"

   4. Under Type, select Email.

   5. Under Send email to, enter the email addresses of the recipients. Use commas to separate multiple addresses.

   6. Enter the details for the email account of the sender.

   7. Click Test to send a test email.

   8. Note

      Some email providers have security filters that prevent users from receiving or viewing large attachments, from receiving scheduled emails and similar. Check the email provider’s security policy to avoid delivery problems and locked email accounts.

   9. Click Save.

5. Set up the automated speaker test:

6. 1. Go to the device interface > System > Events > Rules.

   2. Click Add a rule.

   3. Enter a name for the rule.

   4. Under Condition, select Schedule and select from the trigger list

   5. Under Schedule, select your schedule (“Daily at 6pm”).

   6. Under Action, select Run automatic speaker test.

   7. Click Save.

7. Set up the condition for sending an email when the speaker test fails:

8. 1. Go to the device interface > System > Events > Rules.

   2. Click Add a rule.

   3. Enter a name for the rule.

   4. Under Condition, select Speaker test result.

   5. Under Speaker test status, select Didn’t pass the test.

   6. Under Action, select Send notification to email.

   7. Under Recipient, select your recipient (“Speaker test recipients”)

   8. Enter a subject and a message, and click Save.

This example explains how to set up the audio device to play an audio clip when an Axis network camera detects motion.

• Prerequisites

• The Axis audio device and Axis network camera are located on the same network.

• The motion detection application is configured and running in the camera.

1. Prepare an audio clip link:

2. 1. Go to Audio > Audio clips.

   2. Click > Create link for an audio clip.

   3. Set the volume and number of times to repeat the clip.

   4. Click the copy icon to copy the link.

3. Create an action rule:

4. 1. Go to System > Events > Recipients.

   2. Click + Add recipient.

   3. Type a name for the recipient, for example “Speaker”.

   4. Select HTTP from the Type drop-down list.

   5. Paste the configured link from the audio device in the URL field.

   6. Enter the user name and password of the audio device.

   7. Click Save.

   8. Go to Rules and click + Add a rule.

   9. Type a name for the action rule, for example "Play clip".

   10. From the Condition list, select a video motion detection alternative under Applications.

   11. Note

       If there are no options for video motion detection, then go to Apps, click AXIS Video Motion Detection and turn on motion detection.

   12. From the Action list, select Send notification through HTTP.

   13. Under Recipient, select your recipient.

   14. Click Save.

• This example explains how to:

• Configure DTMF on a device.

• Set up an event to stop the audio when a DTMF command is sent to the device.

1. Go to System > SIP > SIP settings.

2. Make sure Enable SIP is turned on.

   If you need to turn it on, remember to click Save afterwards.

3. Go to SIP accounts.

4. Next to the SIP account, click > Edit.

5. Under DTMF, click + DTMF sequence.

6. Under Sequence, enter "1".

7. Under Description, enter "stop audio".

8. Click Save.

9. Go to System > Events > Rules and click + Add a rule.

10. Under Name, enter "DTMF stop audio".

11. Under Condition, select DTMF.

12. Under DTMF Event ID, select stop audio.

13. Under Action, select Stop playing audio clip.

14. Click Save.

You can set up a rule that plays an audio clip when you receive a SIP call.

You can also set up an additional rule that answers the SIP call automatically after the audio clip has ended. This can be useful in cases where an alarm operator wants to call the attention of someone near an audio device and establish a line of communication. This is done by making a SIP call to the audio device, which will play an audio clip to alert the persons near the audio device. When the audio clip has stopped playing, the SIP call is automatically answered by the audio device and communication between the alarm operator and the persons near the audio device can take place.

1. Enable SIP settings:

2. Go to the device interface of the speaker, by entering its IP address in a web browser.

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

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

5. Click Save.

6. Go to SIP accounts.

7. Next to the SIP account, click > Edit.

8. Uncheck Answer automatically.

1. Play audio when a SIP call is received:

2. Go to Settings > System > Events > Rules and add a rule.

3. Type a name for the rule.

4. In the list of conditions, select State.

5. In the list of states, select Ringing.

6. In the list of actions, select Play audio clip.

7. In the list of clips, select the audio clip you want to play.

8. Select how many times to repeat the audio clip. 0 means “play once”.

9. Click Save.

1. Answer the SIP call automatically after the audio clip has ended:

2. Go to Settings > System > Events > Rules and add a rule.

3. Type a name for the rule.

4. In the list of conditions, select Audio clip playing.

5. Check Use this condition as a trigger.

6. Check Invert this condition.

7. Click + Add a condition to add a second condition to the event.

8. In the list of conditions, select State.

9. In the list of states, select Ringing.

10. In the list of actions, select Answer call.

11. Click Save.

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.

With analytics and apps you can get more out of your Axis device. AXIS Camera Application Platform (ACAP) is an open platform that makes it possible for third parties to develop analytics and other apps for Axis devices. Apps can be preinstalled on the device, available for download for free, or for a license fee.

To find the user manuals for Axis analytics and apps, go to help.axis.com.

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.

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.

AXIS OS determines the functionality of our devices. When you troubleshoot a problem, we recommend that you to start by checking the current AXIS OS version. The latest version might contain a correction that fixes your particular problem.

To check the current AXIS OS version:

1. Go to the device’s web interface > Status.

2. Under Device info, see the AXIS OS version.

1. Download the AXIS OS file to your computer, available free of charge at axis.com/support/device-software.

2. Log in to the device as an administrator.

3. Go to Maintenance > AXIS OS upgrade and click Upgrade.

4. When the upgrade has finished, the product restarts automatically.

If you can’t find what you’re looking for here, try the troubleshooting section at axis.com/support.

When you set up your system, it’s important to consider how different settings and situations affect the required bandwidth (bitrate).

The most important factors to consider:

• Heavy network utilization due to poor infrastructure affects the bandwidth.

• Running multiple AXIS Camera Application Platform (ACAP) applications simultaneously may affect the general performance.

For SD card recommendations, see axis.com.

microSD, microSDHC, and microSDXC Logos are trademarks of SD-3C LLC. microSD, microSDHC, microSDXC 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.

For location of the microphone disable switch, see Product overview.

The microphone disable switch is used to mechanically turn the microphone ON or OFF. The factory default setting for this switch is ON.

RJ45 Ethernet connector with Power over Ethernet (PoE).

VAPIX® is Axis’ own open API (Application Programming Interface). You can control almost all functionality available in Axis devices through VAPIX®. To get access to the complete VAPIX® documentation, join Axis Developer Community at axis.com/developer-community

Enter the commands in a web browser, and replace <deviceIP> with the IP address or host name of your device.

Example

Restart the device

Request

http://<deviceIP>/axis-cgi/restart.cgi

Example

Restore the device. The request returns most settings to default values, but keeps the IP number.

Request

http://<deviceIP>/axis-cgi/factorydefault.cgi

Example

Reset the device. The request returns all settings including IP number to default values.

Request

http://<deviceIP>/axis-cgi/hardfactorydefault.cgi

Example

See a list of all device parameters.

Request

http://<deviceIP>/axis-cgi/param.cgi?action=list

Example

Get a debug archive

Request

http://<deviceIP>/axis-cgi/debug/debug.tgz

Example

Get a server report

Request

http://<deviceIP>/axis-cgi/serverreport.cgi

Example

Capture a network trace of 300 seconds

Request

http://<deviceIP>/axis-cgi/debug/debug.tgz?cmd=pcapdump&duration=300

Example

Enable FTP

Request

http://<deviceIP>/axis-cgi/param.cgi?action=update&Network.FTP.Enabled=yes

Example

Disable FTP

Request

http://<deviceIP>/axis-cgi/param.cgi?action=update&Network.FTP.Enabled=no

Example

Enable SSH

Request

http://<deviceIP>/axis-cgi/param.cgi?action=update&Network.SSH.Enabled=yes

Example

Disable SSH

Request

http://<deviceIP>/axis-cgi/param.cgi?action=update&Network.SSH.Enabled=no