Polycom SoundPoint IP Bat Phone
Tuesday, December 30, 2008
Just so you know - I just finished watching the Dark Night on the old telly. Inspired by what I saw, I decided to write this post.
I'll also mention that this post assumes you have a few things..
1) A Polycom phone
2) Latest BootROM and SIP Firmware
2) a basic working knowledge of the default config files and override files
3) a boot server... TFTP FTP - What ever floats your boat
Disclaimer - This isn't a tutorial.. There is alot here that you'd need to know ahead of time to make this work... My example below only shows you what to change and where- an understanding of Polycom provisioning from a dedicated boot server is required... If you are an existing .e4 client please contact support@e4strategies.com for further documentation.
As you may know, there are many ways to restrict dialing patterns in most modern IP PBX platforms. Often times with these systems, admins can create rules that will disable local, long distance, and even limit the caller's ability to dial anything other than internal extension.
Today's focus will be a feature that forces a Polycom SoundPoint IP phone to dial only one designated number without end-user input from the number pad. Calls to the this number are initiated simply by picking up the handset. Attempts to dial from the key pad will always yield the same result. This feature is much like a hotline phone that you may remember from the analog world where there are no buttons whatsoever, much like commissioner Gordon's "Bat Phone".
In our scenario, the bat phone is used in front of a locked door with a note stating lift handset to dial. Their call will then be connected to a specific person or a call group that will have the ability to let them in or merrily send them on their way. Another application for this feature would be in a grocery store where shoppers can call department managers or workers for assistance without having to read a list of extensions - Dont forget the commissioner's application. A Polycom phone even has a blinking red light when Batman leaves and urgent message-
This feature is simple - Here is what you need to do:
Find the phone1.cfg on your phones boot server. To perform this properly you will need to add the code listed below to your extension's override file. (I have attached examples for your viewing pleasure.) Applying this feature to the stock phone1.cfg will make all phones contacting the boot server perform this same function.
Search for:
Replace with something like this...
autoOffHook call.autoOffHook.1.enabled="1" call.autoOffHook.1.contact="999"
In my example 999 is the number dialed when the handset is lifted.
batphone.zip
I'll also mention that this post assumes you have a few things..
1) A Polycom phone
2) Latest BootROM and SIP Firmware
2) a basic working knowledge of the default config files and override files
3) a boot server... TFTP FTP - What ever floats your boat
Disclaimer - This isn't a tutorial.. There is alot here that you'd need to know ahead of time to make this work... My example below only shows you what to change and where- an understanding of Polycom provisioning from a dedicated boot server is required... If you are an existing .e4 client please contact support@e4strategies.com for further documentation.
As you may know, there are many ways to restrict dialing patterns in most modern IP PBX platforms. Often times with these systems, admins can create rules that will disable local, long distance, and even limit the caller's ability to dial anything other than internal extension.
Today's focus will be a feature that forces a Polycom SoundPoint IP phone to dial only one designated number without end-user input from the number pad. Calls to the this number are initiated simply by picking up the handset. Attempts to dial from the key pad will always yield the same result. This feature is much like a hotline phone that you may remember from the analog world where there are no buttons whatsoever, much like commissioner Gordon's "Bat Phone".
In our scenario, the bat phone is used in front of a locked door with a note stating lift handset to dial. Their call will then be connected to a specific person or a call group that will have the ability to let them in or merrily send them on their way. Another application for this feature would be in a grocery store where shoppers can call department managers or workers for assistance without having to read a list of extensions - Dont forget the commissioner's application. A Polycom phone even has a blinking red light when Batman leaves and urgent message-
This feature is simple - Here is what you need to do:
Find the phone1.cfg on your phones boot server. To perform this properly you will need to add the code listed below to your extension's override file. (I have attached examples for your viewing pleasure.) Applying this feature to the stock phone1.cfg will make all phones contacting the boot server perform this same function.
Search for:
autoOffHook call.autoOffHook.1.enabled="" call.autoOffHook.1.contact=""
Replace with something like this...
autoOffHook call.autoOffHook.1.enabled="1" call.autoOffHook.1.contact="999"
In my example 999 is the number dialed when the handset is lifted.
batphone.zip
Price Drop on Linksys SPA962
Monday, December 29, 2008
In an effort to make our move as easy as possible we are planning a sale to reduce some of the bulge- Our backs will thank us later. 
The first product we have up for grabs is the Linksys SPA962 Just $198.50 - Orders over $1000 USD will also have the added benefit of free ground shipping to the lower 48.
Disable the services/applications button on your Polycom Phone
Sunday, December 28, 2008
I'll apologize in advance for the formatting below (read on you'll see what I mean) Blogger simply refuses to keep the formatting that I have outlined on the "edit post" screen and I am too lazy to tinker with the HTML.
As you may know, higher end Polycom phones have a little button that allows the user to access phone's Mircobrowser. The Microbrowser is like any Web browser—Microsoft Internet Explorer and Firefox, for example—but supports only a subset of XHTML features. It can connect to Web servers hosted in the Internet or intranet and download XHTML pages. The Microbrowser supports a limited number of XHTML 1.0 features—it does not have full Web browser functionality.
The Microbrowser downloads XHTML content from a Web server into the phone’s memory, then parses the content to identify XHTML tags and renders these tags onto the phone’s graphic display. The appearance of the rendered page depends on the graphical capabilities and display size of the device on which the browser is running. Complicated pages should be avoided on devices with very small displays.
The Microbrowser does not support scripting (such as JavaScript). All actions on data entered into forms is processed by the server using POST or GET methods.
For those who have nothing to connect to on the microbrowser the feature can be disabled.
To disable the "Applications" or Services key:On a 550/560/600/601/650/670, Applications/services is key #29
Replace the following in sip.cfg:
<keys key.scrolling.timeout="1" key.x.29.function.prim="Null"/>
With
<keys key.scrolling.timeout="1" key.x.29.function.prim="Null"/>
(Be sure to define the x variable.)
Usable Parameters for x Value...
IP_300
IP_330
IP 430
IP_500
IP_550
IP_600
IP_650
IP_4000
IP_7000
29 represents key #29
Refer to the attached document for key numbers of other models. Default_key_layouts.pdf
Digium Launches ADA - Asterisk Desktop Assistant
Monday, December 22, 2008
This leaves me saying- FINALLY!
The Full Story can be read @ the Digium Blog
http://blogs.digium.com/2008/12/22/asterisk-desktop-assistant-windows-click-to-call-and-more/
The Full Story can be read @ the Digium Blog
http://blogs.digium.com/2008/12/22/asterisk-desktop-assistant-windows-click-to-call-and-more/
Snom 820 Screenshots- WEBUI
Please note: I, like many of my customers will gladly sacrafice call quality for appearances. This is evidenced by the Linksys SPA962 that has found its way happily on my desk for over a year.
Having said that...
The Jury is still out on the snom 820 phone- I can say that from having only made 10 or so calls on the device that I am certainly more impressed with this phone than I have been with all previous snom phones- However, I think that I should use it for a full day before I praise or complain too much.
Here are a few screenshots of the Web User Interface. Click to see...
Having said that...
The Jury is still out on the snom 820 phone- I can say that from having only made 10 or so calls on the device that I am certainly more impressed with this phone than I have been with all previous snom phones- However, I think that I should use it for a full day before I praise or complain too much.
Here are a few screenshots of the Web User Interface. Click to see...
Labels: snom 820
Introducing the Cisco SPA525G SIP Phone with Wifi and Bluetooth Support
Now that Linksys has been officially and completely gobbled up by the networking behemoth Cisco they will be launching new SPA phones under the Cisco brand. Having said that, they are launcing a new IP phone that has features that extend far beyond what I have seen on the road map from other manufacturers.
Enter the 525G - This new IP phone will offer the flexibility of a corded desk set while offering the mobility of a WiFi enabled device. This new phone also supports a few features, that as far as I know are industry firsts, one being support a Bluetooth headset without the need of additional hardware (Correct me if I am wrong) and two being an integrated MP3 player.
SPA525G Info:
The Cisco SPA525G IP Phone is a full-featured VoIP (Voice over Internet Protocol) phone that provide voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
The SPA525G phone is connected to the network through its Ethernet connection or the built-in Wireless-G connection. If you are using the Wireless-G connection, a separate power adapter (PA-100) is required.
The SPA525G provides an additional Ethernet port that allows a computer to be connected to the network through the IP phone. (This option is only available when phone is connected to the network via the wired Ethernet connection).
Unlike traditional phones, the Cisco SPA525G requires a separate power source. Either connect your phone to an Ethernet switch that provides Power over Ethernet (PoE), or use a separate power adapter (PA-100).
• 2 Ethernet 10/100 Mbps ports
• 802.3af Power over Ethernet support
• USB 2.0 host port for connecting a USB memory device to play MP3 music files
• AUX port (to attach a SPA932 attendant console)
• Bluetooth capability for headset support
• 2.5mm stereo earphone jack for headset
• Wireless-G client support
• Kensington security slot support
Order Online @ http://www.8774e4voip.com/Cisco_SPA525G_p/cisco-spa525g.htm
SPA525G Info:
The Cisco SPA525G IP Phone is a full-featured VoIP (Voice over Internet Protocol) phone that provide voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
The SPA525G phone is connected to the network through its Ethernet connection or the built-in Wireless-G connection. If you are using the Wireless-G connection, a separate power adapter (PA-100) is required.
The SPA525G provides an additional Ethernet port that allows a computer to be connected to the network through the IP phone. (This option is only available when phone is connected to the network via the wired Ethernet connection).
Unlike traditional phones, the Cisco SPA525G requires a separate power source. Either connect your phone to an Ethernet switch that provides Power over Ethernet (PoE), or use a separate power adapter (PA-100).
• 2 Ethernet 10/100 Mbps ports
• 802.3af Power over Ethernet support
• USB 2.0 host port for connecting a USB memory device to play MP3 music files
• AUX port (to attach a SPA932 attendant console)
• Bluetooth capability for headset support
• 2.5mm stereo earphone jack for headset
• Wireless-G client support
• Kensington security slot support
Order Online @ http://www.8774e4voip.com/Cisco_SPA525G_p/cisco-spa525g.htm
Labels: cisco spa525g
Digium and Thirdlane partner up to deliver multi-tennant solution.
Wednesday, December 10, 2008
Digium(R), Inc., creator and primary corporate sponsor of the popular Asterisk(R) open source telephony platform, and Third Lane today announced that Third Lane has become a Digium software partner. The companies have collaborated for years and Third Lane has contributed to the open source Asterisk project sponsored by Digium. Today's partnership assures enterprise customers as well as Internet service providers (ISPs), Internet telephony service providers (ITSPs) and resellers that Third Lane solutions are fully compatible with unmodified Asterisk and Asterisk Business Edition(TM).
Digium's Asterisk telephony platform is used on millions of servers worldwide to manage voice over IP (VoIP) calls for businesses and individuals. In addition to creating and maintaining open source Asterisk, Digium offers a full line of products based on the software, including the turnkey Switchvox(R) IP PBX, professional grade Asterisk Business Edition and the open source software appliance AsteriskNOW(TM).
Digium's Asterisk telephony platform is used on millions of servers worldwide to manage voice over IP (VoIP) calls for businesses and individuals. In addition to creating and maintaining open source Asterisk, Digium offers a full line of products based on the software, including the turnkey Switchvox(R) IP PBX, professional grade Asterisk Business Edition and the open source software appliance AsteriskNOW(TM).
"Digium has long benefited from Third Lane's contributions to the Asterisk community, so we're pleased to formally partner and to work closely with each other," said Jim Webster, director of technology partnerships at Digium. "Third Lane's PBX solutions fit a segment of the market looking for a solution that allows significant customization, and companies in the hosting business who need multi-tenant PBX support. When combined with Asterisk Business Edition, Third Lane's products address those customers while providing a commercial-grade telephony platform supported by the creators of Asterisk."
Third Lane provides a family of telephony products based on Asterisk that include advanced management tools designed for use by service providers and businesses. Organizations select the Thirdlane PBX when they require a highly configurable, open system. Significantly, Third Lane is the first company in the Digium ecosystem to provide a true multi-tenant PBX that allows ITSPs and resellers to host virtual PBXs for their customers -- oftentimes small or highly distributed companies. Thirdlane Multi Tenant PBX supports numerous virtual PBXs on a single server, allowing ITSPs to host telephony services by requiring relatively low investment in equipment in order to enter the market.
Erik Smith is chief technology officer of BluegrassNet, a Kentucky-based ISP. BluegrassNet is a Third Lane customer and partner, and a Digium Authorized Reseller. "Hosted VoIP solutions based on Asterisk and the Thirdlane Multi Tenant PBX are popular with our customers because they get all the features of a business phone system with low up-front costs, fast deployment and reasonable monthly payments," Smith said. "Competitive products are very expensive and often charge per seat. We're seeing strong interest in Asterisk-based solutions such as Third Lane's because they're simply more flexible and affordable."
Alex Epshteyn, founder of Third Lane, commented: "Companies work with Third Lane for our proven Asterisk expertise and multi-tenant PBX solution. They recognize and value best-in-class capabilities, so for us, partnering with Digium became a key way to validate our deep knowledge of Asterisk and promote interoperability with Asterisk Business Edition."
About Third Lane
One of the first management tools for Asterisk, Thirdlane PBX simplifies customization and configuration of the Asterisk IP-PBX and is used by businesses and service providers worldwide. Thirdlane PBX's extensive feature set and high-performance design make it an ideal platform for service providers and facility managers that want to offer hosted IP-PBX services to multiple tenants. Because the system delivers easy access to the full power and functionality of Asterisk, system integrators and developers can use it to efficiently build and deploy end-to-end businesses processes that integrate voice. Founded in 2003, Third Lane Technologies, LLC is based in Fairfax, California. For more information please visit www.thirdlane.com or call 1 (415) 261-6600.
About Digium
Digium(R), Inc., the Asterisk(R) Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware to enable resellers and customers to implement turnkey solutions or to design their own voice over IP (VoIP) systems. More information is available at www.digium.com.
$6 Off Polycom's Productivity Suite - CODE = VUCPS
Tuesday, December 9, 2008
Just in time for whatever Holiday you might be celebrating...
Order Here Just enter VUCPS on the checkout page-
Polycom Productivity Suite Video:
Order Here Just enter VUCPS on the checkout page-
Polycom Productivity Suite Video:
XO's SIP and IP Flex gets the green light from Digium
XO Communications and Digium, the Asterisk Company, today announced that Digium’s Switchvox IP PBX has been certified to work with XO’s award-winning Session Initiation Protocol trunking solution, XO SIP. The certification will streamline the adoption of XO SIP by users of the award-winning Switchvox telephony system and offer them a cost-effective and efficient way to connect to the public phone network without having to rely on traditional land lines.
Today’s announcement begins a collaboration between two leading providers of VoIP services and solutions for businesses. Today, more than 15,000 businesses and more than 500,000 customer employees benefit from VoIP services offered by XO through its portfolio of converged services, which includes XO SIP and XO IP Flex.
Digium’s Asterisk is the world’s most widely used open source telephony software, and Switchvox is the most widely used customer premise-based IP PBX based on Asterisk. “Businesses are looking for both high-performance and cost-effective VoIP solutions that are easy to deploy and manage,” said Nicola Jackson, director of voice and converged services for XO Communications.
“Together, Digium’s feature-rich Switchvox platform and XO SIP offer an affordable, high-quality VoIP solution that businesses can deploy rapidly. Completing interoperability with Digium Switchvox platform enables us to offer businesses more options when selecting their SIP trunking services provider.”
The certification confirms that Switchvox and XO SIP work seamlessly together, providing businesses with confidence when using their feature-rich, easy-to-use, inexpensive Switchvox phone system with XO SIP.
“Companies that want to integrate voice and data with Switchvox are now assured of full interoperability when they select XO for their IP network,” said Leslie Conway, vice president of global marketing for Digium. “SMBs and solution providers want certified solutions that have been thoroughly tested. Formal XO certification now provides them with a high level of confidence in choosing Digium and XO as a combined solution for their network needs.”
Today’s announcement begins a collaboration between two leading providers of VoIP services and solutions for businesses. Today, more than 15,000 businesses and more than 500,000 customer employees benefit from VoIP services offered by XO through its portfolio of converged services, which includes XO SIP and XO IP Flex.
Digium’s Asterisk is the world’s most widely used open source telephony software, and Switchvox is the most widely used customer premise-based IP PBX based on Asterisk. “Businesses are looking for both high-performance and cost-effective VoIP solutions that are easy to deploy and manage,” said Nicola Jackson, director of voice and converged services for XO Communications.
“Together, Digium’s feature-rich Switchvox platform and XO SIP offer an affordable, high-quality VoIP solution that businesses can deploy rapidly. Completing interoperability with Digium Switchvox platform enables us to offer businesses more options when selecting their SIP trunking services provider.”
The certification confirms that Switchvox and XO SIP work seamlessly together, providing businesses with confidence when using their feature-rich, easy-to-use, inexpensive Switchvox phone system with XO SIP.
“Companies that want to integrate voice and data with Switchvox are now assured of full interoperability when they select XO for their IP network,” said Leslie Conway, vice president of global marketing for Digium. “SMBs and solution providers want certified solutions that have been thoroughly tested. Formal XO certification now provides them with a high level of confidence in choosing Digium and XO as a combined solution for their network needs.”
Labels: switchvox