I had to write up something for a newsletter that tried the gpedit.MSC approach.  That way had many pitfalls and several limitations.  I gave them a simple regfix to do the same thing.

Stop CDs from Auto Running:
If you don’t want your CDs or DVDs to Autorun, you can turn this off.

To do this in any system from 95 to Vista:

To re-enable Autorun:

With that done I launched the admin software. The phone system is listed by mac address and model #. I chose our system and pressed connect. There is another option to specify the IP as well. This opens up another screen and requires the needed info. I went the easy route, after all this needs to be simple enough to be installed by the end user. After pressing connect you get an “Are you sure this is your system?” type prompt. Agree and enter the default password of admin. Note this is case sensitive. It will ask you to change your password and log you in.

Setting up a phone was very easy. You connect it to the network and click phones on the menu. Choose add new phone. The phone will appear in a list of unassigned phones. Choose your phone and then choose assign user. This brings up a new screen. One of the cool things I noticed about this was I can assign the phone to a person, location, group, or Job Role. This will change the identifier (CID) of the phone. I chose location and put in Front Desk as this will be my Operator/Secretary phone. Choose next and it programs the phone and brings the phone up as its new extension.

All in all it is very easy! This may be a competitor to companies like the one I work for. If you have a small business customer who needs a phone system you can ship this to them and they can figure 99% of it out. I don’t see that as being a good solution as you want to build a relationship with the customer. I see easily see this system being sold at Wal-Mart and Best Buy. It has a basic user set with a few bells and whistles. It includes the vital services such as Voice mail and address book integration as well as cool features as dial by voice. If I want to call the front desk for example I press the blue “rp” button and say call “bob”. The system repeats back basically what you said and then connects the call.

I am going to play and will write more later.

Cheers!

17041

Released on 2009-04-06

Enhancements

• Unified Communications
  
  • Fax - Send and receive faxes using Switchvox. You can use your fax machine, or fax files from your desktop computer.
    
  • Video calling - You can now use video phones with Switchvox. Video phones do not require any special set up, but for video quality we recommend that you use the same kind of phone on both ends of the call.
    
  • Chat - The new Switchvox private chat server uses the XMPP protocol. The Switchboard offers a Chat Panel, or you can use your favorite XMPP-based client. Also, you can set the chat (Jabber) hostname for Switchvox in Network Settings, and the chat (Jabber) hostname for any VOIP Providers that run a Jabber-based chat server that you will use (including a peered Switchvox).
    
  • IMAP Mailbox - Each extension now has a one-stop IMAP Mailbox for both voicemail and faxes.
    
• Admin Features
  
  • Phone Setup for snom technology, Inc. phones - Phone Setup is available for VOIP phones manufactured by snom technology, Inc. (Phone Setup for Polycom, Inc. phones has been available since v9337).
    
  • Disk-space Quotas - You can set a disk-space quota for voicemail and fax usage for each extension and extension group. Also, you can see the voicemail and fax disk-space usage, in total and for each extension.
    
  • Call queue improvements - Queue members can log into, out of, and pause their status on each queue (rather than a set of queues). In the Switchboard, queue members can pause their status (when they are logged in), and add a comment about why they are unavailable or when they might be back. Also, queue status changes are now available as URL Manager events.
    
  • Notification Templates - Admins can customize multiple templates for voicemail notifications. These templates are available to users as ‘default’ templates. When Users sets up their notifications, they can select from these default templates or (if they have permission) customize their own templates.
    
  • Centralized Voicemail on Peered Switchvoxes - You can set up a peered Switchvox as the SIP Provider for External Voicemail.
    
  • Bulk Import Extensions tool - You can use a CSV file to create and modify extensions.
    
  • IVR Menu Actions -
    
    • Send an email (with or without a sound)
      
    • Set global (system-wide) variables
      
    • Get global (system-wide) variables
      
    • Get extension status
      
    • Get extension type
      
    • Check a user password
      
    • Perform math
      
    • Concatenate variables
      
    • Send a recorded sound to a voicemail box
      
    • Store a recorded sound in the Sound Manager
      
    • Upload a sound to a Web application
      
    • Play a sound from a web application
      
    • Merge sounds
      
  • Access Control - You can set up multiple rules to limit network access to Switchvox services.
    
  • BRI: B410P Support - This card is configured in the Channel Admin.
    
  • HD Voice G722 Support - High quality wideband audio.
    
  • 6-digit extensions - Extensions can be 3, 4, 5, or 6 digits.
    
  • Scheduled Reports - You can set up regularly scheduled reports to be emailed as HTML, XML, or a chart.
    
  • Advanced reports on inbound DIDs - Call Reports can now be created based on the Incoming DID.
    
  • Network and VOIP Provider Settings -
    Network Settings: IP ToS settings (audio and video)
    VOIP Providers: RTP Port Range
    SIP provider:
    
    • Caller-ID method
      
    • Include user=phone in SIP
      
    • Use Local Address in From Header
      
    • Enable Jitterbuffer
      
    • Allow Reinvite
      

    
    Channel Admin:
    

    • Ring Debounce (for analog lines with FXO signaling)
      
    • Network Specific Facility Code (for PRI T1/E1 Bearer channels)
      
    • PRI Reset Interval (for PRI T1/E1 Bearer channels)
      
  • Password-strength indicator - When an Admin or a User is creating a new password, a password strength-indicator lets him know how strong his new password is.
    
  • Subscription-Expiration Alerts - New alerts let you know if your Subscription is expiring soon (or if it has already expired), and Renewal links help you easily get your Subscription renewed.
    
  • XML API: Switchvox Extend - An XML-based API lets administrators access call logs, call reports, and extension lists.
    
  • Digium Addon Products - Admins can easily register Digium products for Switchvox. In particular, you can register your fax license (in the Admin Suite, select Machine Admin -> Digium Addon Products).
    
  • Advanced Debugging - You won’t need to use these tools unless requested by a Switchvox Service Representative. It includes SIP and PRI debug, and Asterisk console viewer.
    
• User Features
  
  • Voicemail Greetings - You can customize multiple voicemail greetings, specify which greeting should be active, and assign different greetings to play when you are unavailable versus on the phone.
    
  • Organized Phonebooks - You can create and manage multiple phonebooks, to keep your contacts organized. You can also see each contact’s `Additional Numbers’ that they publish, and create and modify your own additional numbers for them.
    
  • Additional Numbers - You can set up your own `Additional Numbers’ so that co-workers can easily call you at the numbers you publish (your mobile phone, or an extension on the manufacturing floor).
    
  • Call queue improvements - You can log into, out of, and pause your status on each queue. When you pause your status, you can add a comment to let people know when you might be back.
    
  • Conference Announcement - A caller can record his name before entering a conference room. Then, when he enters and exits the conference room, the recording is played.
    
• Switchboard
  
  • Chat Panel - The chat panel includes the company directory for easy communications.
    
  • Company Directory Panel - The Company Directory panel offers type-to-find, to help you quickly find your co-workers’ extensions.
    
  • Multiple Phonebook Panels - Each of the your Phonebooks is available as a Switchboard panel.
    
  • Centralized presence - Across peered Switchvox PBXs, Phonebook entries show consistent Presence and call details.
    
  • Presence Settings - Presence settings include:
    
    • Available
      
    • Away (includes a comment)
      
    • Chatty
      
    • Extended Away (includes a comment)
      
    • Do Not Disturb
      
  • One-click options - These features are now available with quick one-click access in your Switchboard panels:
    
    • Phonebook entries let you dial additional phone numbers.
      
    • Extended entries let you Barge and Whisper.
      
    • In/out/pause queue-status icons let you set your status for each queue (including a comment for a paused status)
      

Changes

• The extension type `Call Queue Agent’ has been removed. If you have extensions of this type, you can still maintain them. However, we recommend using a `Virtual’ extension instead, which offers the same features as the `Call Queue Agent,’ but without the limitations.
  

Here are the ports we need for the Alitgen VOICE.

Voip – RTP   -  UDP 49152-49998

IP/IPtalk – tcp 10032,10064 - UDP 10060 – UDP 30000-31000 – UDP 10064

I did a bit of research and it does not appear that you can enter ranges via the web interface.  So here is the script you need to have this program up the ports via the command line.

On a desktop/laptop on the network.

Start – Run – telnet (enter the IP of the switch) - enter.

You should see the a similar line as this:  ( the HP 5406 part can be anything)

HP 5406 Switch#

Copy all of the following to the clip board from conf to exit.

Conf

qos udp-port range 49152 49998 priority 7

qos udp-port range 30000 31000 priority 7

qos tcp-port 10032 priority 7

qos tcp-port 10064 priority 7

qos udp-port 10064 priority 7

qos udp-port 10060 priority 7

wr mem

exit

Right click on the top of the dos window and choose edit – paste.  It will run all of the commands and save the changes.

You should end up on the same command line as where you started.

Move on to the next switch.

Good luck!

The D-Link Response Point

I have noticed something that is glaringly missing from this kit. POE. All devices have power cords so keep that in mind when you install it. I would recommend a small UPS for each desk. Otherwise your phones go dead when the power is out. I think that the addition of POE would make this a much better kit as you would need fewer UPS units and fewer cords at the desktops. Moving on…

After looking at the system I am wondering how necessary the switch will be. The Router has an 8 port on it and should handle a small phone system fine. We shall see.

I am going to set the system with just the Router (Left), Phone System (right), and one Phone (center). I will also connect my laptop to the setup to run the CD that came with the system.

I am still very pleased with the exceedingly small size of this system!

To get the unit going I powered up the router for IP, the phone system, and then a phone. So far it is very plug and play. The phone, phone system, and my laptop pulled an IP with no issues. All of the devices came with a short cat 5 cable. I am a little concerned the length of these as they appear to be about 4 ft long. I have a hard time getting a 7 ft cable to be long enough on most installs. I would recommend purchasing a 7ft plus cable for each phone before you install this.

I was just handed a D-LINK DVX-2000MS Response Point phone system bundle to play with. (GOODY!) I started opening boxes and this is what I found: A D-link switch, a Business D-link router, 5 D-Link phones, an analog gateway, and the phone system.

To start with the phone system is very small. (4″X9″X8″ approx) It looks like a micro PC that has most of the connectors blocked off. USB, audio out, a 12v barrel connector, and a NIC are the only ports available. I can see that if limited space is an issue this will be a good option.

The switch (DES-1228) and router (DIR-130) are from D-Links business line. I do not have much experience with these but I am pleased they did not include the Best Buy consumer models as they are not all that great. It also had a wireless router (DIR-330) with it that I will not use in this test.

The phones (DPH-125MS) are small with few buttons. They have a basic LCD and use a menu system for some features. I am very happy to see hold, mute, and voicemail have physical buttons. I have dealt with soft buttons and such for those and that makes it a pain to use so good job there. It has a speaker phone as well. They are also nice looking to boot.

The Analog Gateway is something I am happy to see for several reasons.  First is expansion.  The Old system was what you see is what you get.  With the migration to gateway devices you can “stack” the units to add lines as you need them within reason.  The second reason is that you can have your analog gateways near the dmark which is generally speaking a hostile environment.  This leaves the system to be placed next to a managers desk or next to the server if you have one.  That way if you damage the gateway you can just buy a new one as opposed to having to replace the whole system.  Very nice!

Now we have the hardware listed I will start the setup but that is the next post.

Cheers!

Simply add a clock to any location on the network via RJ-45 connection. Power to the clock is provided by Power-Over-Ethernet (PoE) which eliminates the need for additional costly outlets.

Consistent and accurate time across an entire organization maintains efficiency and smooth company operation:

Manufacturing Facilities – Synchronized time promotes adherence to schedules keeping the organization consistently meeting productivity goals.

Healthcare - Daily healthcare requires synchronized time schedules which are critical for patient care and timely delivery of medications.

Schools & Universities - Accurate timekeeping is crucial in transitioning students from one class to the next. Access control systems (security) require synchronized timing … doors lock at a certain time all across the campus.

Valcom IP Clock Features:

Set automatically by NTP (Network Time Protocol)

10/100BaseT Ethernet compatible

DHCP or static IP addressable

Power over Ethernet (IEEE 802.3af)

No master clock required

Simple User Control for:

IP configuration

NTP server address

Time zone offset from UTC, Daylight Savings Time

Time Display, 12/24 hour format

Status reporting

Industrial grade metal housings

Digital Clocks

2.5” or 4.0” numerals with four (4) or six (6) digit display

Red LED seven-segment display

Analog Clocks

12” or 16” round

Valcom’s IP PoE Clock Systems eliminate the need for specialized software – manage one single clock or thousands of clocks from any PC on the network.

For additional information on Valcom’s IP PoE Clock Systems

I should give you a little back story…

In July my company started selling a phone system from Digium called SwitchVox. I was very excited about this product as it had TONS of features an innovative web interface and very easy configuration. The other major plus is that almost everything was in the box ready to go. No additional licenses for IVR (auto attendant), Call groups, voicemail, etc. If Switchvox could do it, it was free with the system. (there are some add-ons from other vendors you have to get separately but these are things like voice recognition and other cool options.) SwitchVox is an awesome package for a very reasonable price. We tested it for some time and it seemed to work great in house. Ok, let’s sell it as it looks great! First install went smoothly and it all seems ok. It was a very quick install and a very happy install tech. Next day, Customer calls the echo is terrible to the point of not useable, odd. Lets run the built in echo settings tool. It seems better now. It was rinse and repeat after that as the issue kept returning. We called tech support. They were Johnny on the spot and great to work with. SwitchVox configured up a phone remotely so they could hear the issues and worked to resolve them. It was better now. A month later the issues retuned. This is getting old. Now keep in mind we have several installs out now and they all had the same issue. Strange that our in house demo unit did not have the issue. Contacted the supplier (Interlink Web) and they were the ones to tell me the ZAPTel driver in the software did not properly support the newly released analog cards. Of course that is what we sold out customer, the latest and greatest. Sigh.

Now I am still waiting as they want to get the release “just right” and support a bunch of new features. From what I hear the new features are holding up the release. Just one thing, I COULD CARE LESS ABOUT NEW FEATURES AT THIS POINT! I just want the box to work with analog lines. It appears that they are more worried a about video phones (cool) working then making the existing systems function. Now that we are near the end and the release is near I was able to hold on to most the customers and keep them on SwitchVox. Now reach install is different. One is on a SIP trunk so they do not use the analog lines and is ecstatic about the system another other wants the echo problem to just get fixed.

Don’t get me wrong I like SwitchVox a lot and will continue to recommend them once they fix the analog issue. As far as features, options, and configuration goes it is a great solution. They just need to work on problem resolution over the latest new feature.

Just the latest rant.

Cheers!

——–

Can you pull the little black cord from the phone wait 1 min. and then plug it back in for me?

——–

OK did that now I have a dial tone.  I tried that yesterday but it did not work so glad this time it did.

Thanks for your help!

——–

No problem. Unfortunately you may have problems now and again as the phone is going through a VPN. Usually the best fix is as follows.

Open a Command window (Start – run –Type CMD – enter)

A black window should appear. In that black window type Ping 192.168.40.10.

You should get a response back with times.

If you get a “Timed Out” comment give the office a call or wait 10 min and try again.

If it pings back properly; please pull the power from the phone for one min then plug it back in.

——–

It seems these kinds of procedures are necessary for older proprietary VOIP systems. The newer SIP based solutions (Asterisk/Switchvox/Trixbox) seem to have an easier time of it as does the Altigen VOIP as it seems to have been designed with this in mind. In time as the bandwidth and quality on the internet improves I think many companies will not even have a phone system as it will all be “hosted” elsewhere and the phones will just connect via the net. Until that day if you have remote users take a hard look at Altigen and Astrisk to control all of just your remote users or deal with the randomness of the VPN tunnel.

Cheers!