Avaya just released the R9.1 firmware version for their popular IP Office platform. What’s new you might ask?
Here is a basic overview of the changes being made:
- New Premium Offering
- IP Office Server Edition Improvements
- System expansions and increased resiliency
- IP Office When Deployed as a Branch Improvements
- Aura System Manager and central management
- Aura centralized applications, services, and solutions
- Branch VM Pro and centralized management
- IP Office Unified Communications and Video
- New mobile VoIP client enhancements
- Lync Plugin enhancements
- Exchange 2013 integration
- Avaya Communicator (Flare) enhancements
- IP Office Key Feature Improvements
- New system capacity expansion
- SIP service provider (SIP Trunk) features
- SSL/VPN remote access and continued IPOSS support improvements
- Communications accessibility
- Call center enhancements
- IP Office Web Manager Evolution
- Expanded integration of management capabilities
- Sever Edition and Standard Editions
- IP Office Security Enhancements
- TLS and SRTP support for SIP traffic
- Encryption of all H.323 traffic via Line, Trunk, and SCN
IP Office R9.1 will be supported on the following platforms:
- IP500 V2
- IP Office for Linux (Server Edition, Virtualized Server Edition, Select Edition)
Some major changes that may affect people wanting to upgrade to R9.1:
- Discontinued support for: IP401, IP403, IP406, IP406 V2, IP412, IP500 V1, or Small Office Edition platforms
- Discontinued support for: IP400 Trunk Cards, IP400 VCM Cards, IP400 Digital Station V1, IP400 Phone V1, IP400 So8, and IP500 Legacy Card Carrier
- Discontinued support for: Customer Call Reporter (CCR/Advanced Edition). Existing customers will need to migrate to the new IP Office Contact Center Solution.
For more information, see General Availability (GA) of IP Office Release 9.1 for details.
The Avaya B179 IP Conference Phone works perfectly with Avaya IP Office. It is supported on IP Office R5.0 and higher, and requires an Avaya IP Endpoint license. We will provide a basic tutorial here to get your Avaya B179 IP Conference phone connected. Please see the Avaya B179 Install and Admin Guide for a more in-depth guide.
1) First, let’s make sure the IP Office is setup correctly:
- Log into your IP Office
- Go to the License Tab > Make sure you have an Avaya IP Endpoint license available (the 3rd Party IP/SIP Endpoint will not work)
- Go to System > LAN1 or LAN2 (whichever interface is connected to the network the B179 is on) > VoIP tab:
- Check “SIP Registrar Enable”
- Create a new extension by selecting Extension > create SIP Extension
- Add a valid Base Extension number i.e. 8000
- Create a new user by selecting User > create User:
- Add a valid Name ie Extn8000
- Add the Base Extension number that was previously created. i.e 8000
- Go to the Telephony tab > Supervisor Settings tab:
- Put a numerical password in the “Login Code” field. For testing purposes we can use 8000.
2) Then, configure the Avaya B179:
- On the B179 conference phone:
- Press Menu > Status > Network
- Set or Confirm the IP Address of the phone (DHCP/Static)
- This IP address is used to log into the web interface of the phone.
- Web Settings:
- Enter the IP address of the phone into a web browser of your choice
- Login with the default credentials
- User: Admin
- Password: 1234
- Settings > SIP
- Enable Account 1
- Account Name: This is what is shown on the display of the B179.
- User: IP Office Base Extension number i.e. 8000
- Registrar: IP of LAN1 or LAN2 IP Office interface (whichever is used)
- Realm: *
- Authorization Name: IP Office Base Extension number i.e. 8000
- Authorization Password: Enter the “Login Code” setup for the user.
- Save values
- Settings > Networking
- These fields are used to make sure your B179 is setup correctly to the network
- IP Address: IP address of B179
- Net Mask: Subnet mask of the network on which the B179 is connected
- Gateway: Local gateway on the network on which the B179 is connected
- DNS info can be manually specified
- Save values
- Reboot the phone and the B179 should register with the IP Office
This should get your Avaya B179 IP Conference Phone connected to your IP Office for testing purposes. Then from here on, you will want to fine tune your settings and secure your phone following the Avaya B179 Install and Admin Guide.
Polycom makes some of the best quality SIP VoIP Phones on the market. They work well with most IP-PBX and Hosted VoIP Providers. While most manufacturers and providers will have their own setup guidelines (which you should follow first), there are times when you may need to manually configure your Polycom VoIP phone. We will provide a very basic guide on getting your Polycom phone manually registered to an Asterisk SIP-based system here.
First you will need some basic information to register your Polycom phone:
- IP address of the Polycom phone
- IP address of the Asterisk system and SIP Port used to register (usually 5060)
- SIP extension number and phone password from your Asterisk system for your Polycom Phone
Once you have obtained the information, open up your web browser and enter in the IP Address of the Polycom phone in the Address Bar. Once it prompts you to login, use the following user name and password (assuming it is still default):
- Username: Polycom
- Password: 456
When logged into the Polycom phone’s user interface go to the Lines section (may be under Settings on newer firmwares) and Line 1. Enter the information into the following fields:
- Display Name: SIP Extension Number assigned on the Asterisk system ie. 2000
- Address: SIP Extension Number assigned on the Asterisk system ie. 2000
- Label: SIP Extension Number assigned on the Asterisk system ie. 2000
- Authentication User ID: SIP Extension Number assigned on the Asterisk system, i.e., 2000
- Authentication Password: Phone-password assigned on the Asterisk system
Under Server 1:
- Address: IP Address of the Asterisk system
- Port: Port number for SIP registration ,i.e., 5060
- Leave all other settings default.
Submit to Save settings, which should prompt you to reboot the phone. Check to see if the Polycom phone is registered properly by pressing on the phone: Menu/Settings > Status > Lines.
You should be all set!
With the ever increased threat of hacking, it is very important to follow proper steps in securing your public facing devices. Breaches can occur from multiple fronts, i.e., allowing any SIP and VoIP traffic in your system, improper firewall configurations, and of course not changing default passwords. With Avaya IP Office, there are several steps in making sure your system is locked down as much as possible. Avaya has issued a technical bulletin on doing this with the following basics:
Review your existing installations and/or new deployments. Determine any security risks and requirements then implement these changes:
- Change security defaults and passwords
- Remove any unnecessary accounts
- Disable any unused services/interfaces
- Enforce strict password policies
- Make sure users and extensions are secure
- Make sure trunks/lines are secure
- Prevent and block unwanted Calls
- Secure user voicemail and one-X Portal accounts
- Block unnecessary and limit IP network exposure
- Make sure management applications & configuration data are secure
- Make sure servers running IP Office applications are secure
- Activate reporting/monitoring of your system
- Test for vulnerabilities
- Install latest software updates/service packs
Even with these steps taken, you will want to continually monitor your alarms and logs to detect any unusual activity. Always keep up with security advisories and make sure to keep your system up-to-date with the latest patches and upgrades.
Please read the document, Securing your IP Office Guidelines, for full detail on securing your IP Office.
There are some situations where you need to have an analog phone line where there are no existing cable runs except for a data network. Or you may simply want to extend analog phone line(s) to a remote site. No worries! We have a solution for you that utilizes Grandstream SIP VoIP ATA’s and Gateways in Peer-to-Peer mode. There are 3 different scenarios with appropriate solutions:
1) Peer-to-Peer using ATA’s only. Click Here for the solution using Grandstream HT502 and HT503’s.
2) Peer-to-Peer using ATA’s and Gateways. Click Here for the solution using Grandstream GXW’s and HT50x’s.
3) Peer-to-Peer using Gateways only. Click Here for the solution using Grandstream GXW’s.
These are 3 different examples of extending your analog lines (POTS) over a data network utilizing Grandstream’s cost effective solutions. If you have any questions, feel free to contact any one of our The Telecom Spot representatives at 866-369-3394.
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Konftel adds Microsoft Lync compatibility to one of their most versatile conferencing devices, the Konftel 55 and Konftel 55W.
If you havent purchased one yet, ask one of our knowledgeable representatives at 866-369-3394. Buy yours today here at The Telecom Spot: Konftel 55, Konftel 55W
Konftel 55 Series – The hub of your communications Computers, smartphones, tablets and desktop phones – the Konftel 55 series is designed to keep pace with the array of devices used for communicating in the conference room today.
Layer for Lync
The user interface have a new and fresh design. Features that are supported and displayed on the Konftel 55 series display screen:
- answer/end call
- call information; name, subject and talk time are displayed
- your status information (presence)
- last number redial
- keypad for calling via Lync – mute – voice mail
The Panasonic SIP IP Phones are top-notch end points from a brand name you trust. Getting them setup and working usually isn’t very difficult if you know what info needs to go into what fields. The following tutorial applies to the follow phones, however, some interfaces may change slightly from model to model: KX-UT113-B, KX-UT123-B, KX-UT133-B, KX-UT136-B, KX-UT248, KX-UT670, KX-TGP500, KX-TGP550.
First you will need to setup the extension on your Asterisk System (the process can vary depending on the distribution). From this process, you will need the:
- IP Address of the Asterisk Server: ie: 192.168.42.100
- SIP Port of the Asterisk Server: ie: 5060
- Extension SIP ID/Alias: ie: MAC Address of phone
- Extension Number: ie: 7000
- Extension Password: ie: 1234pass
- Voicemail Access Number: ie: 8500
Next we will get started on configuring the phone itself. Please note, this is tutorial on how to get your phone connected. Advanced functionality is not covered:
- Connect the phone to the network and power it on
- Once connected to the network, you will need to turn on embedded web from the individual phone settings (turns off after a while). Press the Setup button > Select Embedded web > Select On
- Open up a web browser of your choice and type in the IP Address of the Panasonic IP Phone into the address bar. A login screen should appear. The default login is L: admin P: adminpass.
- Once logged in, you should make sure you have upgraded the firmware to the current version, the link to the admin guide and firmware files can be found here: http://www.panasonic.net/pcc/support/sipphone/download/us.html
- Select the VoIP tab and click on Line 1 under SIP Settings. Fill in these fields:
- Phone Number: Extension # assigned in Asterisk. ie 7000
- SIP URI: Extension SIP ID/Alias assigned in Asterisk. ie: MAC Address of phone
- Registrar Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
- Registrar Server Port: SIP Port of the Asterisk Server. ie: 5060
- Proxy Server Address: IP Address of the Asterisk Server. ie: 192.168.42.100
- Proxy Server Port: SIP Port of the Asterisk Server. ie: 5060
- Presence Server Adress: IP Address of the Asterisk Server. ie: 192.168.42.100
- Presence Server Port: SIP Port of the Asterisk Server. ie: 5060
- Outbound Proxy Server Address: IP Address of the Asterisk Server: ie: 192.168.42.100
- Outbound Proxy Server Port: SIP Port of the Asterisk Server: ie: 5060
- SIP Authentication ID: Extension SIP ID/Alias on the Asterisk. ie: MAC Address of phone
- SIP Authentication Password: Extension Password on the Asterisk. ie: 1234pass
Lastly, Select the Telephone tab. Then Select Line 1 under Call Control. Fill in these fields: Voice Mail Access Number: Voicemail Access Number on the Asterisk: ie: 8500
Click Save and reboot the phone.
Your Pansonic IP Phone should be configured to work on your Asterisk IP PBX. Any question? Just leave a comment or contact us at The Telecom Spot for assistance.
Often the first steps to troubleshooting a Polycom IP phone is to make sure you have an up-to-date or the correct firmware loaded. If you are trying to upgrade the firmware or factory default one or two Polycom phones, then the easiest way is to use Polycom’s provisioning server:
- Go to http://voipt2.polycom.com and follow their directions.
If you are working with multiple phones or need more control over the config files, the easiest to way upgrade is through a local TFTP server:
- First off you need a TFTP server, also known as, Trivial FTP server. You can find many programs on the internet but the one that we use in house PumKIN (PumpKIN Can be downloaded at: http://kin.klever.net/pumpkin/ )
- The next step would be to gather all of the files you need to update your Polycom device. You can check the compatibility of the firmware to your device and also download most releases at the Polycom SIP Software Compatibility Matrix. Some of the latest firmware will only be available through us directly. Please email email@example.com with your order number to receive the instructions for downloads.
- Create a folder called TFTP and place it on your C: Drive. Unzip the downloaded files into this directory making sure the bootrom and sip files are listed in root of the C:\TFTP directory
- Open PumpKIN up and click on Options. Under the Server tab, change the TFTP file-system root to C:\TFTP and Read Request Behavior to Give all Files. Press OK and make sure all software firewalls are turned off or configured to allow access to PumpKIN on the host computer.
Now that your TFTP server is running, we need to make sure the Polycom Unit is set up to search for TFTP updates. On the unit itself (This process might be slightly different depending on the phone):
- Make sure the phone is connected to the same network as the TFTP server.
- Power on the phone and press the SETUP button when prompted.
- Enter the password of 456
- Scroll down to Server Menu and select
- Change the server type to Trivial FTP
- Change server address to the IP Address of TFTP Server you created (the IP Address of the PC running PumpKIN)
- Exit the menu, and the phone will prompt you to Save and Reboot, select OK.
Upon reboot, the phone will go through the boot process and display Uploading Bootrom on the screen. It will reboot a couple of times after updating the rest of the files. When the phone completes the process, it will display a welcome screen, which shows the current firmware version loaded.