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!
Are you currently using a Cisco IP phone system? Cisco is discontinuing their UC solution for SMB customers and have declared it end of life (EOL) for nearly a year. Their EOL announcement included Cisco’s UC300 and UC500 series; the support entitlements for these platforms will begin phasing out January 1, 2015.
For many SMB customers, this is a great time to start replacing those Cisco phone systems with a Digium Switchvox solution – which can often be purchased for less than the annual cost of a Cisco support contract! From now thru
September 30 December 31, 2014 Digium will offer additional discounts up to 15% off MSRP to help you make the switch. Simply contact one of our helpful representatives here at The Telecom Spot to start!
- Digium promotion is for qualified product purchases July 1 – September 30, 2014 by end user customers only.
- Customers must request this promotional discount at time of order placement as neither Digium nor The Telecom Spot can provide promotional discounts after orders have been invoiced
- Only qualified Switchvox products are eligible for the promotional discount. This include Switchvox appliances, Switchvox SMB software, Switchvox subscriptions. A complete list of qualifying products and part numbers are listed below.
- This offer does NOT apply to cold spares, updates and maintenance, subscription renewals, phone feature packs, extended hardware warranties, phones, Digium cloud services, or any other items that are not included on list of qualifying products
- This promotion applies only to new Switchvox solutions replacing existing Cisco telephony solutions. Digium may contact customer to confirm the replacement for their promotional claim audits.
Eligible products for the Cisco Replacement Promotion
|Digium Part Number / SKU
||Switchvox 80 Appliance, NA
||Switchvox 80 Appliance, AU
||Switchvox 80 Appliance, EU
||Switchvox 80 Appliance, UK
||Switchvox 310 Appliance, NA
||Switchvox 310 Appliance, AU
||Switchvox 310 Appliance, EU
||Switchvox 310 Appliance, UK
||Switchvox 360 Appliance, NA
||Switchvox 360 Appliance, AU
||Switchvox 360 Appliance, EU
||Switchvox 360 Appliance, UK
||Switchvox 380 Appliance, NA
||Switchvox 380 Appliance, AU
||Switchvox 380 Appliance, EU
||Switchvox 380 Appliance, UK
||Switchvox 450 Appliance, North America
||Switchvox 450 Appliance, Australia
||Switchvox 450 Appliance, Europe
||Switchvox 450 Appliance, United Kingdom
||Switchvox 470 Appliance, North America
||Switchvox 470 Appliance, Australia
||Switchvox 470 Appliance, Europe
||Switchvox 470 Appliance, United Kingdom
||Switchvox SMB Software Download
||Switchvox SMB DVD with Reg Code
||1 Switchvox Gold Subscription for 1 User
||5 Switchvox Gold Subscriptions for 5 Users
||25 Switchvox Gold Subscriptions for 25 Users
||100 Switchvox Gold Subscriptions for 100 Users
||1 Switchvox Platinum Subscription for 1 User
||5 Switchvox Platinum Subscriptions for 5 Users
||25 Switchvox Platinum Subscriptions for 25 Users
||100 Switchvox Platinum Subscriptions for 100 Users
||1 Switchvox Titanium Subscription for 1 User
||5 Switchvox Titanium Subscriptions for 5 Users
||25 Switchvox Titanium Subscriptions for 25 Users
||100 Switchvox Titanium Subscriptions for 100 Users
From now thru
Dec 31, 2013 March 31, 2014 get free Switchvox User Training ($995 value) with purchase of any eligible Digium Switchvox system or software. The Digium Switchvox training, Introduction to Switchvox: User Interface On-Line Training – English, is ideal for new users that want to learn how to get started using their Switchvox business phone systems.
Eligible Switchvox models include:
Please Note: The free user training offer is only available at the time of a qualified purchase, and will not be available separately post-purchase. Contact us for details.
Grandstream has just released their new UCM6100 series IP PBX with no licensing fees, bringing enterprise-grade communication solutions to SMBs at affordable entry-level price point! Available in 4 model sizes, Grandstream’s UCM6100 series delivers advanced voice, video, data and mobility applications with a single-server solution. They’re available now at The Telecom Spot at limited quantities – remember to add-to-cart to view our current special pricing!
Grandstream UCM6100 Series IP PBX
- Powered by advanced Asterisk based open source platform
- Supports up to 500 users and up to 60 concurrent calls
- Affordable pricing for SMB budgets – no licensing fees
- 4 model sizes ranging from 2 to 16 FXO ports
- 2-year manufacturer warranty
- No licensing/recurring fees per features like IVR, auto attendant
- Free lifetime firmware updates
- Auto discovery and registration of SIP endpoints (phones, cameras, etc)
- Zero-configuration provisioning via intuitive WEB UI
- Plug and play setup and remote maintenance
- Broad SIP interoperability with popular service providers, SIP trunk providers and other SIP hardware, including Grandstream GXP phones and GXP video cameras.
Enterprise-Grade features for SMBs:
- Voice - Customizable features like Caller ID, voicemail, auto attendant, IVR, DND, MWI, call routing, click-to-dial, 3-way conferencing and more; comprehensive codec support
- Data – Call detail records; Fax; Corporate phonebook LDAP files and servers; call recording; voicemail/fax to email and more
- Video – Real-time SIP video calling; video camera/surveillance integration; 2-way audio from cameras; extensive video codec
- Mobility – Remote workers can make/receive calls on smartphones and laptops; extension dialing over all locations; access files and monitor business remotely across the globe
|PSTN Line FXO Ports
|Analog Telephone FXS Ports
||Single or Dual (UCM6102 only) 10M/100M/1000M RJ45 Ethernet port(s) with integrated PoE Plus (IEEE 802.3at-2009)
||No No No
||Power/Ready, Network, PSTN Line, USB, SD
||128×32 graphic LCD with DOWN & OK button
Digium just released their best PBX cards for any Digium installation with the TE133 and TE134 digital telephony cards. They are designed as replacements to the TE121BF and TE122BF cards and offer an incredible introductory price of $495 (log into our site for even better pricing) and include echo cancellation and the single span card.
- Up to 24 (T1/J1) or 30 (E1) simultaneous calls
- Selectable T1, E1 or J1 Mode / Half-Length, Half-Height, Digital Card
- PCI-Express (TE133) or PCI (TE134)
- One (1) RJ48 Interface Port
- Protocol support includes: ISDN PRI, Robbed-Bit, CAS, MFC/R2
- Built-in 128ms Octasic DSP hardware echo cancellation
- 5 year manufacturer warranty
The TE133 and TE134 cards support industry standard telephone protocols including Primary Rate ISDN (North American and Euro standard) protocol families for voice. Both line-side and trunk-side interfaces are supported as well as advanced call features. Octasic based hardware echo cancellation is built into the cards to eliminate the task of echo cancellation from systems CPC which increases overall system performance and call quality. They are supported with both Switchvox and Asterisk, plus a 5 year manufacturer warranty from Digium.
Both the Digium TE133 an TE134 digital telephony cards are now in stock and ready to ship at The Telecom Spot! Initial stock is limited though, so don’t wait too long to get yours.
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.