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.