Configuration For Phones

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1 POWERED BY GTI NETWORK Configuration For Phones Instruction on how to set up different type of phone and ATA s Instruction on how to set up different type of phone and ATA s

2 Contents Top Page... 5 Aastra 6730i/6731i VoIP Phone... 5 Overview... 5 Configuration Details... 6 Atcom AG188N... 8 Configuration Details... 9 Configuration Screen... 9 Cisco Linksys PAP Configuration Details Configuration Screens How to avoid the long delay to hear the ringtone Cisco Linksys PAP2T Configuration Details Configuration Screens Customer Submitted Information How to avoid the long delay to hear the ringtone Linksys PAP2T ATA Adapter Reset Procedure Cisco SPA Configuration Details Cisco Linksys SPA942 NA Configuration Details See also What is a Dial Plan? Digit Sequence Examples See also References Cisco IP Phone 7940/ Configuration Details Cisco SPA2100 Phone Adapter Configuration Details... 34

3 How to avoid the long delay to hear the ringtone Cisco SPA2102 Phone Adapter with Router Configuration Details How to avoid the long delay to hear the ringtone Dial Plan for Linksys ATAs Dial Plan Digit Sequence Examples See also References Cisco SPA504G Phone Configuration Details Grandstream HandyTone Configuration Detail Message Waiting Indicator Grandstream HandyTone Configuration Detail Grandstream HandyTone Configuration Detail Mediatrix 4100 Series Configuration Details Optional Setting: Netgear WGR615V Configuration Detail OBi Configuration details Default Dial Plan: Panasonic KX-TGP Configuration Details Pirelli DP-L Configuration Details Polycom SoundStation IP 4000 Conference Phone... 63

4 Configuration Details Configure Settings Polycom SoundPoint IP 501, 550, 650, etc Configuration Details Configure Settings Siemens Gigaset C450-Ip Configuration Details SNOM 320 VoIP Phone See Configuration Details Using the "Retrieve" Voice mail Key Snom m3 VoIP Phone Configuration Details Telco AC Configuration Details Yealink SIP-T28P (VSRF) Configuration Details Zycoo ZP Configuration Details BASIC SETTING Polycom IP450 Setup & User Guide SoundPoint IP 450 HD Quick User Guide V Key Location Directories More Detailed Information from the Manufacturer Manually configuring a Polycom SoundPoint IP 320, 321, 330, 331, 450, 550, 560, 650, 670 and SoundStation IP 6000, 7000 with 3CX Phone System Polycom SoundPoint IP 335 HD Voice Quick User Guide KEY LOCATIONS Provisioning a Polycom SoundPoint IP 320, 321, 330, 331, 335, 450, 550, 560, 650, 670 and SoundStation IP 5000, 6000, 7000 with 3CX Phone System Set up instructions Cisco SPA301-G1 User Guide... 90

5 Phone Layout Setup detail Cisco SPA303G User Guide Phone Layout Setup detail Top Page Aastra 6730i/6731i VoIP Phone Aastra 6730i VoIP Phone Product: Aastra 6730i/6731i VoIP Phone Company: Aastra Overview: The Aastra 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML

6 capability to access custom applications and is fully interoperable with leading IP-PBX platforms. Supported by a host of Aastra configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications. Configuration Details Get the IP Address Press the Spanner key Press the scroll down button to get to Phone Status and then press [ confirm selection Select IP&MAC Addresses and press [ ] The IP Address will appear on the screen. ] to Web Management Open your web browser and go to the IP you just found: https://[deviceip] You will be presented with a warning; you need to confirm that you want to continue. You will be asked for user and password. Provide the user and password for the phone o Username: Admin o Password: xxxxx On the left menu select Line 1 under Advanced Settings Fill in the values as shown in the image.

7 Screen Name: John Smith (value of your choice) Phone Number: (Your GTI username) Caller ID: (Your GTI username) Authentication Name: (Your GTI username) Password: xxxxxxxxxxx (the account password) Proxy Server: sip.ncsvoice.com (one of our multiple servers)* Proxy Port: 5060 (default sip port) Backup Proxy Port: 0 Outbound Proxy Port: 0 Registrar Server: sip.ncsvoice.com (one of our multiple servers)* Registrar Port: 5060 (default sip port)

8 Backup Registrar Port: 0 *According to some customers, using the server and the port number under Registrar Server help them obtaining registration with our server. For example, sip.ncsvoice.com :5060. Click Save Settings. Restart the Phone. Click on Reset under Operations on the left menu. Now click on Restart Phone will restart automatically Top Page Error! Bookmark not defined. Atcom AG188N Atcom AG188N Product: Atcom AG188N

9 Company: Atcom Overview: AG188N voice gateway is an Internet based one port voice gateway. AG188N ATA adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service. AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches. Configuration Details SIP Configuration Section: Fill the following fields according to your account: o Register Server Addr: sip.ncsvoice.com (one of our multiple servers) o Register Server Port: 5060 o Register Username: (your GTI username) o Register Password: ******** (the account password) o Detect Interval Time: 60 o DTMF Mode: DTMF_RFC2833 o Enable Via rport: Enabled o Local SIP Port: 5060 o Register Expire Time: 60 o RFC Protocol Edition: RFC3261 o Server Type: common o SIP Default Protocol: Enabled Click on the "Apply" button at the bottom of the form. Configuration Screen

10 Atcom AG188N Configuration Screen Top Page Error! Bookmark not defined.

11 Cisco Linksys PAP2 Cisco Linksys PAP2 Product: Linksys PAP2 Company: Cisco Overview: The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a softswitch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface. Each analog telephone is presented as a distinct SIP user. Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following: Dial: **** (That is 4 asterisks) Once this is done, dial: 110# (110 followed by a square) The system should now playback the IP Address your device has been assigned. (Example: )

12 Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter. (example Replace by the IP Address your device is currently using. Step 3 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 4 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com (You can choose any of our multiple GTI servers) Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO DNS SRV Auto Prefix: NO Step 5 (Optional) Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. At the bottom of Line 1 TAB, you will find a field called Dial Plan

13 Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 6 Click on the "Save Settings " button at the bottom of the form. Step 7 Switch to the SIP tab and scroll down to RTP Parameters and set the follow setting: RTP Packet Size: Configuration Screens PAP2 Configuration Screen

14 PAP2 Configuration Screen

15 PAP2 Configuration Screen Customer Submitted Information: For North America: Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with GTI Network. Read the article called: Configure your Linksys VoIP ATA the right way! How to avoid the long delay to hear the ringtone If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2. Changing the PAP2's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting: Note: However before changing that option, test if calling the number with an # at the end of the number works(e.g #). If that doesn't work you need to contact the support staff in voiptelprovider.com 1- First access the PAP2's web interface. 2- Click on the Admin Login and then click on (switch to advanced view) 3- Click on the Regional tab and look for the Control Timer Values (sec) section. 4- Enter the desire value in the Interdigit Long Timer field (for example lower this value to 4).

16 Note: There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET# command with these boxes as you may be locked out of your device's settings by doing so. Top Page Error! Bookmark not defined.

17 Cisco Linksys PAP2T Cisco Linksys PAP2T Product: Linksys PAP2T Company: Cisco Overview: The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet. With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voic , call forwarding, distinctive ring, and much more. See Configuration Details with Static IP If you are running the PAP2t with a tomato firmware router, this may represent a persistent issue after

18 rebooting the device, generally it won't register back as usual, in order to fix this you only need to set Tomato UDP Unreplied timeout down to 10 (from default of 30). Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following: Dial: **** (That is 4 asterisks) Once this is done, dial: 110# (110 followed by a square) The system should now playback the IP Address your device has been assigned. (Example: ) Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter. (example Replace by the IP Address your device is currently using. Step 3 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 4 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com (You can choose any of our multiple GTI servers) Register Expires: 180 Proxy Fallback Intvl: 180

19 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO DNS SRV Auto Prefix: NO Step 5 (Optional) Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. At the bottom of Line 1 TAB, you will find a field called Dial Plan Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 6 Click on the "Save Settings " button at the bottom of the form. Configuration Screens

20 PAP2T Configuration Screen PAP2T Configuration Screen

21 PAP2T Configuration Screen Customer Submitted Information: For North America: Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with Smart Voice Read the article called: Configure your Linksys VoIP ATA the right way! To upgrade a firmware version from a Windows system, the PAP2T-NA documentation nor Cisco's web site does not say how to do this. Go to Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports. Click on the link "Download Firmware", which downloads the.zip file. Run the.exe file, then enter the IP number of the ATA device (called SPA in the program). It then upgrades the device. This is provided for your information: the author is not saying you have to upgrade the firmware.

22 How to avoid the long delay to hear the ringtone If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2T. Changing the PAP2T's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting: Note: However before changing that option, test if calling the number with an # at the end of the number works(e.g #). If that doesn't work you need to contact the support staff in Smart Voice. 1- First access the PAP2T's web interface. 2- Click on the Admin Login and then click on (switch to advanced view) 3- Click on the Regional tab and look for the Control Timer Values (sec) section. 4- Enter the desire value in the Interdigit Long Timer field (for example lower this value to 4).

23 Linksys PAP2T ATA Adapter Reset Procedure Sometimes it will be very helpful to reset your linksys ATA adapter to factory default settings # Connect a telephone to line 1 of the PAP2T unit and power it on. Disconnect your PAP2T adapter from the internet connection(unplug the Ethernet cable from the PAP2T hardware unit). Resetting with internet connection may mess up the unit making it completely useless. Dial ****, and wait for the Interactive Voice Menu (IVM) to get activated. Type in the following number including the # symbol. (This number spells RESET.) Confirm this by pressing 1. Your linksys ATA unit will now go back to it factory default settings. Note: There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET$ command with these boxes. Top Page Error! Bookmark not defined.

24 Cisco SPA112 Cisco SPA112 Product: SPA112 Company: Cisco Overview: The Cisco SPA112 2 Port Adapter enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections. The Cisco SPA112 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. Each phone line can be configured independently. With the Cisco SPA112, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines as well as control their migration to IP voice with an extremely affordable, reliable solution. Make sure to install the latest firmware from Version 1.1 or later should be used for proper Caller ID support. Configuration Details Step 1 Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following: Dial **** from the phone, even though there is no dial tone. When you hear "System Configuration Menu," dial # slowly. The current IP address will be read back. If you hear , check your network connection and DHCP server. If necessary, a static IP address

25 can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad (for example, 10*1*27*2 for ). The network mask can be set with option 121# and the default gateway can be sent with option 131# Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/doc-9900 document. Step 2 Open your web browser and go to the IP address you obtained in step 1 (for example, The default username is admin, and the default password is also admin. Step 3 Go to Quick Setup and configure Line 1 as follows: Proxy: sip.ncsvoice.com Display Name: Your name User ID: Your GTI SIP Account number (or subaccount) Password: Your GTI SIP Password Dial Plan: (911S0 <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) (note: replace 555 in the dial plan with your area code) Click Submit to save settings. Step 4

26 Click on Voice, then Line 1 Set NAT Mapping Enable to Yes, then set NAT Keep Alive Enable to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its' traffic will not be subject to NAT in this configuration. Under Proxy and Registration set Register Expires to 180, Proxy Fallback Intvl to 180 Also confirm the following settings: Use DNS SRV: NO DNS SRV Auto Prefix: NO Click Submit to submit these changes Step 5 Click Network Setup, then go to Basic Setup, then click Time Settings Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct. Click Submit to save the changes

27 Cisco Linksys SPA942 NA Cisco Linksys SPA942 NA Product: Linksys SPA942 NA Company: Cisco Overview: Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines, or can be configured to use a shared number over multiple phones. Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for easy menu- and web-based configuration. Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following: Dial: **** (That is 4 asterisks) Once this is done, dial: 110# (110 followed by a square) The system should now playback the IP Address your device has been assigned. (Example: )

28 Note: Some multi-line Cisco IP 'phones do provide a [setup] key (icon is one piece of paper with one corner folded). On these models, pressing [setup] then selecting 'Network' from the menu provides its IP without dialing any codes. The remainder of the configuration process remains the same. Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter. (example Replace by the IP Address your device is currently using. Step 3 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 4 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO DNS SRV Auto Prefix: NO Step 5 (Optional) Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

29 At the bottom of Line 1 TAB, you will find a field called Dial Plan Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 6 Click on the "Save Settings " button at the bottom of the form. See also Dial Plan for Cisco and Linksys devices; the same rules apply to dial plans for Cisco's IP 'phones as to Linksys ATA's. What is a Dial Plan? The dial plan is a string of characters that determine how the digits input in your keypad are interpreted and transmitted by your ATA device. And also determine if the number dialed is accepted or rejected. This way, you can use a dial plan to facilitate dialing and also block certain types of calls (either long distance or international). Note: Please notice, that you can also block the international calls in your VoIP.ms account. Digit Sequence A dial plan contains a series of digit sequences, separated by the character and the entire set of sequences is enclosed within parentheses. Each time you press a key in your keypad your ATA device is going to try matching that key with each digit sequence in your dial plan. Digit Sequence * # x [sequence] Function You can use any of these characters to represent a key pressed in your keypad. This represent any character on the phone. You can enter characters between brackets to create a list of accepted digits. For example, if you enter [1-5] this allow the user to press any digits from

30 1 to 5. You can also create a list using numbers along with other characters, for example [35-8*] allows the user to press either 3, 5, 6, 7, 8 or *.. (period) <dialed:substituted> You can use a period to accept zero or more entries of a give digit. For example, 01. allows the user to enter 0, 01, 011 and so on. This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The dialed digits can be zero or more characters. For example with this sequence <:1555>xxxxxxx if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press , the system transmits This can be used between digits to play an outside line dial tone after a user-entered sequence., (comma) For example, with this sequence 9, 1x. an outside line dial tone is sounded after the user presses 9, and the tone continues until the user presses 1 You can use this character to prohibit a dial sequence.! (exclamation point) For example with the sequence 1900xxxxxxx! the system reject any sequence that starts with S0 or L0 This override the setting in the Short inter-digit timer or Long inter-digit timer to 0 seconds. P# (where # is the duration of This provide with a pause given the amount of seconds. the pause in seconds) Examples Here's a few examples of digit sequences that you can add to your dial plan. To dial any international number without using the 011 prefix. <:011> [2-9]xxxxxxxx. You can also accomplish this if you set the Dialing Mode to E164 in your Account Settings To block a call to an specific area code (replace 555 with the area code you want)

31 <:1> 555 xxxxxxx! The next sequence, allows you to dial your Phone book entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520 <:*75>xx< # : > See also ATA's IP Phones References Cisco SPA2100 Phone Adapter Cisco SPA2102 Phone Adapter with Router Cisco Linksys PAP2 Cisco Linksys PAP2T Cisco SPA300/500-series 'phones Cisco Linksys SPA942 NA Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones Top Page Error! Bookmark not defined. Cisco IP Phone 7940/7960

32 Cisco IP Phone 7940/7960 Product: Cisco IP Phone 7940/7960 Company: Cisco Overview: The Cisco IP Phone 7940/7960 is a full-feature telephone that provides voice communication over an IP network. This phone functions like a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Configuration Details 1. Press **# to unlock the Configuration Protection (some versions can also be unlocked pressing the "settings" button, scroll all the way to the bottom to the "unlock phone" option and then press the "select" softkey). 2. Press Settings button 3. Go to "Network Configuration" to set the IP connectivity 4. Go to SIP Configuration 5. Select Line 1 In Name and Authentication Name enter your GTI username(e.g ) Enter your GTI password Enter any of the GTI which is sip.ncsvoice.com as Proxy Address Configure Preferred Codec to g711u Press Back and then Save to apply changes. As an additional recommendation, Set "NAT enabled" to Yes, and on the "NAT address" parameter, enter the IP address of Router or firewall if, any.

33 Top Page Error! Bookmark not defined. Cisco SPA2100 Phone Adapter Cisco SPA2100 Phone Adapter Product: Cisco SPA2100 Phone Adapter Company: Cisco Overview: Inexpensive, easy to install, and simple to use, the Cisco SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet. The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

34 Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following: Dial: **** (That is 4 asterisks) Once this is done, dial: 110# (110 followed by a square) The system should now playback the IP Address your device has been assigned. (Example: ) Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter. (example Replace by the IP Address your device is currently using. Step 3 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 4 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password)

35 Use DNS SRV: NO DNS SRV Auto Prefix: NO Step 5 (Optional) Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. At the bottom of Line 1 TAB, you will find a field called Dial Plan Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 6 Click on the "Save Settings " button at the bottom of the form. How to avoid the long delay to hear the ringtone If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting: Note: However before changing that option, test if calling the number with an # at the end of the number works(e.g #). If that doesn't work you need to contact the support staff in Smart Voice 1- First access the SPA's web interface. 2- Click on the Admin Login and then click on the Voice tab. 3- Click on the Regional tab and look for the Control Timer Values (sec) section. 4- Enter the desire value in the Interdigit Long Timer field (for example lower this value to 4).

36 Note: The image correspond to the PAP2 device. If your device doesn't have this setting, contact GTI support. Top Page Error! Bookmark not defined. Cisco SPA2102 Phone Adapter with Router Cisco SPA2102 Phone Adapter with Router Product: Cisco SPA2102 Phone Adapter with Router Company: Cisco Overview: Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet. The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

37 Configuration Details This is a guide for the Initial Configuration of the SPA2102. Step 1 Power off your network devices, including your modem and PC. Connect an Ethernet Cable from the Ethernet port of the SPA to the Ethernet port of the PC. Connect an Ethernet Cable from the Internet port of the SPA to the LAN/Ethernet port of the Modem. Connect your regular handset phone, to the Line port of the SPA2102. Then power up modem, then the SPA2102 and then the PC. Launch a web browser from the PC and enter "http:// /advanced" in the URL address bar field. After these steps you should now have access to the Web Interface page of the SPA2102 to start with the initial configuration. NOTE: If this is not working or the Browser can't find the page, you may also need to enable the Administration web service. Dial 7932#, then when prompted press 1 to enable and 1 to confirm Step 2 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 3 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password)

38 Use DNS SRV: NO DNS SRV Auto Prefix: NO Step 4 (Optional) Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. At the bottom of Line 1 TAB, you will find a field called Dial Plan Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 5 Click on the "Save Settings " button at the bottom of the form. How to avoid the long delay to hear the ringtone If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting: Note: However before changing that option, test if calling the number with an # at the end of the number works(e.g #). If that doesn't work you need to contact the support staff smart Voice 1- First access the SPA's web interface. 2- Click on the Admin Login and then click on the Voice tab. 3- Click on the Regional tab and look for the Control Timer Values (sec) section. 4- Enter the desire value in the Interdigit Long Timer field (for example lower this value to 4). Note: The image correspond to the PAP2 device. If your device doesn't have this setting.

39 Additionally, whenever the registration is dropped and its having problems to gain it back, you can try this simple trick:on Advanced view, go to the Line tab, look for the SIP port setting, and change it to another, i.e. if you have port 5060 change to 5080, if it is 5061 you can change to Dial Plan for Linksys ATAs The basic dial plan provided in the configuration samples for the Linksys ATA devices (like PAP2, PAP2T and SPA2100), should work without any issue with Smart Voice dial plan: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Note: Replace 555 by the area code of your choice. If you want to have the 7 dial for your area code. This guide has been created in order to help you learn more about the Dial Plan and also you can customize it according to your preferences. Please note that customizing your dial plan is optional. Dial Plan The dial plan is a string of characters that determine how the digits input in your keypad are interpreted and transmitted by your ATA device. And also determine if the number dialed is accepted or rejected. This way, you can use a dial plan to facilitate dialing and also block certain types of calls (either long distance or international).

40 Note: Please notice, that you can also block the international calls in your VoIP.ms account. Digit Sequence A dial plan contains a series of digit sequences, separated by the character and the entire set of sequences is enclosed within parentheses. Each time you press a key in your keypad your ATA device is going to try matching that key with each digit sequence in your dial plan. Digit Sequence Function * # You can use any of these characters to represent a key pressed in your keypad. x This represent any character on the phone. [sequence] You can enter characters between brackets to create a list of accepted digits. For example, if you enter [1-5] this allow the user to press any digits from 1 to 5. You can also create a list using numbers along with other characters, for example [35-8*] allows the user to press either 3, 5, 6, 7, 8 or *.. (period) You can use a period to accept zero or more entries of a give digit. For example, 01. allows the user to enter 0, 01, 011 and so on. <dialed:substituted> This is used for sequence substitution, you can use this to indicate that certain numbers dialed are replaced by other characters. The dialed digits can be zero or more characters. For example with this sequence <:1555>xxxxxxx if the user dial a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user press , the system transmits This can be used between digits to play an outside line dial tone after a user-entered sequence., (comma) For example, with this sequence 9, 1x. an outside line dial tone is sounded after the user presses 9, and the tone continues until the user presses 1 You can use this character to prohibit a dial sequence.! (exclamation point) For example with the sequence 1900xxxxxxx! the system reject any sequence that starts with 1900.

41 S0 or L0 P# (where # is the duration of the pause in seconds) This override the setting in the Short inter-digit timer or Long inter-digit timer to 0 seconds. This provide with a pause given the amount of seconds. Examples Here's a few examples of digit sequences that you can add to your dial plan. To dial any international number without using the 011 prefix. <:011> [2-9]xxxxxxxx. You can also accomplish this if you set the Dialing Mode to E164 in your Account Settings To block a call to an specific area code (replace 555 with the area code you want) <:1> 555 xxxxxxx! The next sequence, allows you to dial your Phone book entries using an speed dial like the POTS provider's. For example if you dial 20# the system will send *7520 <:*75>xx< # : > See also IP Phones References Cisco SPA300/500-series 'phones Cisco Linksys SPA942 NA Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones Top Page Error! Bookmark not defined.

42 Cisco SPA504G Phone Cisco SPA504G Phone Product: Cisco SPA504G Phone Company: Cisco Overview: The Cisco SPA504G uses standard encryption protocols to perform highly secure remote provisioning and unobtrusive in-service software upgrades. Remote provisioning tools include detailed performance measurement and troubleshooting features, enabling network providers to deliver high-quality support to their subscribers. Remote provisioning also saves service providers the time and expense of managing, preloading, and reconfiguring customer premises equipment. Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following: Dial: **** (That is 4 asterisks) Once this is done, dial: 110# (110 followed by a square) The system should now playback the IP Address your device has been assigned. (Example: )

43 Note: Some versions of Cisco IP 'phone (typically multi-line devices from the SPA30x and SPA50x series) provide a [setup] key (icon is one piece of paper with one corner folded). On these models, pressing [setup] then selecting 'Network' from the menu provides its IP without dialing any codes. Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter. (example Replace by the IP Address your device is currently using. Step 3 You should now see the web interface of your Linksys/Sipura. click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View". Step 4 Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: Yes Nat Mapping/Traversal: Yes Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO DNS SRV Auto Prefix: NO (On multi-line 'phones like the SPA30x/50x this tab is labelled 'Ext1', 'Ext2', 'Ext3'. The initial defaults assign all individual line keys to 'Ext1' settings. Labels are specified on the 'Phone' tab) Step 5 (Optional)

44 Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice. At the bottom of Line 1 TAB, you will find a field called Dial Plan Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab (or Ext1 tab) at the bottom of the page: (911S0 310xxxx <:1555>[2-9]xxxxxx 1[2-9]xx[2-9]xxxxxxS0 [2-9]xx[2-9]xxxxxxS0 *xx *xx. [3468] [2-9]xxxxxx 4xxx **275x. xxxxxxxxxxxx.) Step 6 Click on the "Save Settings " button at the bottom of the form. The IP 'phone may take up to 35 seconds to re-initialize Top Page Error! Bookmark not defined. Grandstream HandyTone 286 Grandstream HandyTone 286 Product: Grandstream HandyTone 286 Company: Grandstream

45 Overview: Grand stream s award-wining HandyTone-286 is innovative Analog Telephone Adaptor that offers a rich set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market Configuration Detail Step 1: Click on Account 1 to configure your first line. Step 2: Fill the followings fields. SIP Server: sip.ncsvoice.com (one of our multiple servers) SIP User ID: (Your GTI username) Authenticate Password: ********* (Account Password) User ID is phone number: No SIP Registration: Yes Register Expiration: 180 NAT transversal : No o Notes: NAT transversal can be set to Yes if you are behind a router (even if you do not set a value in the STUN server field). Indeed according to the reference guide, with no specified STUN server, then the phone will only periodically (every 20 seconds by default) send a blank UDP packet (with no payload data) to the SIP server to keep the mapped port open on the NAT. Home NPA: Empty (Some tests show that this field should be kept empty or you will encountered difficulties with some numbers ending with a busy signal [tested with 418 in home NPA, and a number] ) Message Waiting Indicator To enable this feature with your Grandstream HandyTone 286 you need to associate a voic with the account (or subaccount) you have registered in the device and set the SUBSCRIBE for MWI option to YES Top Page Error! Bookmark not defined.

46 Grandstream HandyTone 486 Grandstream HandyTone 486 Product: Grandstream HandyTone 486 Company: Grandstream Overview: Grand stream s award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultra affordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market. Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo. Configuration Detail Step 1: Click on Account 1 to configure your first line. Step 2:

47 Fill the followings fields. SIP Server: sip.ncsvoice.com (one of our multiple servers) SIP User ID: (Your GTI username) Authenticate Password: ********* (Account Password) User ID is phone number: No SIP Registration: Yes NAT transversal : 10 Top Page Error! Bookmark not defined. Grandstream HandyTone 502 Grandstream HandyTone HT502 Product: Grandstream HandyTone HT502 Company: Grandstream Overview: The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 100Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices. In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features. Enhanced security Automated provisioning using symmetric and asymmetric voice Support for a broad range of popular voice codec

48 Configuration Detail Step 1: Click on Account 1 to configure your first line. Step 2: Fill the followings fields. SIP Server: sip.ncsvoice.com (one of our multiple servers) SIP User ID: (your GTI username) Authenticate Password: ********* (account password) User ID is phone number: No SIP Registration: Yes NAT transversal (STUN): No Top Page Error! Bookmark not defined. Mediatrix 4100 Series Mediatrix 4102 Product: : Mediatrix 4102S and 4100 Series running DGW 2.0 firmware Company: Media5 Corporation

49 Overview: The Mediatrix 4100 Series device is a security ready VoIP gateway, connecting up to two analog phones and/or faxes, as well as a PC or a home router to a broadband modem. The Mediatrix 4102 offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling and media transmission aspects. Configuration Details Step 1 The first step is to find out what IP Address your adapter is currently using. By default, the Mediatrix 41xx (4102S, 4104, 4108, 4116 or 4124) running DGW 2.0 gets its IP address by DHCP. If the Power LED is solid on, this means the device has an IP address. You can then go to Step2. If the Ready LED is blinking, the Mediatrix unit is looking for a DHCP server, you can then connect the unit to a network with DHCP server or do a Partial Reset. To discover your device s IP address, pick a phone connected on Line 1 and do the following: Dial: *#*0 The system should now playback the IP address your device has been assigned. (Example: ) Step 2 Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.(example: Replace by the IP address your device is currently using. You should now see the web interface of your Mediatrix unit. The default username is Public The Passowrd :<Empty> (There is no password by default).

50 Step 3 Click on the SIP Menu Sub-menu Servers Registrar host: sip.ncsvoice.com (Replace with the address of one of the multiple servers from Smart Voice) Proxy Host: sip.ncsvoice.com (Replace with the address of one of the multiple servers from Smart Voice) Click Submit Step 4 Click on the SIP Menu Sub-menu Registration Username: (Replace this with your GTI username) Friendly Name: John Smith (Replace with your name or company name) Select Enable in the Register drop-down menu Note: Depending on the model and the number of lines, you may configure from 1 to 24 ports. Click Submit

51 Step 5 Click the SIP Menu Sub-menu Authentication Click Edit on the first row

52 Apply to: Endpoint Endpoint: Choose the FXS port for which you want to configure the Authentication Validate Realm: disable Username: (Replace this with your GTI username) Password: ******** (Type the GTI account/sub-account password) Click Submit & Refresh Registration Optional Setting: Step 6 Click the Telephony Menu CODECS Sub Menu Disable G.711 a-law for both Voice and Data Next to G.729 click Edit and change the Voice priority to 10 Click Submit

53 Step 7 To support User name with _ Click the Call Router menu Sub-menu Auto-Routing Change the Criteria Type to SIP Username Click Apply Config Step 8 If your DHCP server does not provide an SNTP Server, you can configure it manually. Click the Network Menu and then sub-menu Host Change the SNTP Configuration Source to Static Configure the SNTP Host to: time.nrc.ca or any other SNTP server. Click Apply Config At the end of the configuration, if you see the message Some changes require to restart Click on services table and restart all Required Services Top Page Error! Bookmark not defined.

54 Netgear WGR615V Netgear WGR615V Product: WGR615V Company: Netgear Overview: The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold. It is a wireless ( G) router that happens to have a built-in ATA. It is possible to use just the ATA part. Configuration Detail Go to Complete the following fields: SIP Proxy: Your GTI server which is sip.ncsvoice.com SIP Control Port: 5060 or Line Enable: Yes Display Name: whatever you want call display to show

55 Telephone Number: GTI username User Name: GTI username Password: GTI password Register Expire Time (sec.): 30 Registration Head Start Time (sec.): 3 Anonymous Incoming Call: Allow Distinctive Ring Valid: Prefer Ring ID: Prefer Fax Codec: Prefer Codec: Support Codec: Note: special characters such as "_", "#" and possibly others cannot be included in the phone number, username or password fields. Where sub-accounts contain an underscore, you can only use this device with your main GTI account. Top Page Error! Bookmark not defined.

56 OBi110 OBi110 Product: OBi110 Company: OBIHAI Technology Inc Overview: With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet. The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get. Configuration details Dial ***1 from the connected phone, to get the IP address of your device.

57 Service Providers >> ITSP Profile >> SIP Please note that in order to change the settings, you need to Uncheck the Default box on the right hand side. Proxy Server: GTI (one of our multiple servers) Proxy Server Port: 5060 Registrar Server: GTI (one of our multiple servers) Registrar ServerPort: 5060 Service Providers >> ITSP Profile >> General Name: (Your VoIP.ms username) Voice Services >> SP Service AuthUserName: (Your GTI username) AuthPassword: ********* (Account Password) Checking your Voic If you encounter any issues when you try to check your GTI Voic . Please try the following suggestions. Try changing your Dial plan. Replace the 555 digits in the following lines by the area code of your choice and copy the line, including parenthesis, in the Digitmap field in the ITSP Profile: Default Dial Plan: ((1xxxxxxxxxx <1555>[2-9]xxxxxx <1>[2-9]xxxxxxxxx 011xx. xx. (Mipd) Dial Plan to allow you to dial *97 and *98: (1xxxxxxxxxx <1555>[2-9]xxxxxx <1>[2-9]xxxxxxxxx 011xx. xx. *xx. (Mipd) Also, some clients have been successful by dialing **1 and *97 for line 1 or **2 and *97 for line 2.

58 An additional note regarding Outbound Calling In at least one instance it was necessary to specify a non-default outbound calling route in the Obi110 to be able to place calls using the GTI service. The default setting had the Obi110 attempting to place calls using the PSTN port on the device. The relevant setting is: Physical Interfaces >> PHONE Port

59 PrimaryLine: (Select from drop-down) The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups. Top Page Error! Bookmark not defined. Panasonic KX-TGP 550 Panasonic KX-TGP 550 Product: Panasonic KX-TGP 550 Company: Panasonic

60 Overview: Panasonic KX-TGP 550 SIP Cordless Phone System allows you to have up to eight (8) phone numbers. You can set up in several ways: for example, you can set the phone number for each handset. Or you can group the handsets by group setting and restrict the incoming calls receivables to the specific handsets. Handsets if you need them. Support for 3 simultaneous network conversations CODEC: G.711a-law / G.711μ-law / G.722(wideband) / G.729a / G.729(32K) DECT radio technology 2.1" Large LCD with white backlight on cordless handset Up to 6 DECT cordless handsets*1 Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions) Hands-free speaker phone on cordless handset Wall mountable base unit Configuration Details Complete the following fields: Phone number ID: (Replace with your GTI username, or sub account login) Line ID: Leave this field blank. Registrar Server Address: sip.ncsvoice.com (You can replace with any of the multiple GTI servers) Registrar Server port: 5060 (Default SIP Port) Proxy server Address: sip.ncsvoice.com (You can replace with any of the multiple GTI servers) Proxy server port: 5060 Presence server port: 5060 SIP service domain: sip.ncsvoice.com (You can replace with any of the multiple GTI servers) SIP source port: 5060 Authentication ID: (Replace with your Smart username or sub-account login) Authentication password: ********* (Enter Smart Voice s main account/sub account Password) The default registration expiry time on this device is 3600 seconds. You may have to reduce that if you lose registration. Unfortunately, that can not be set on the web interface. You need to load a provisioning configuration file from a web or FTP server (details here, page 132 for file format). The parm is REG_EXPIRE_TIME_[n]="mmm", where n is line number, and mmm is the registration interval. e.g. REG_EXPIRE_TIME_1="180" to set line 1 registration time to 180 seconds. A symptom of this problem is that the telephone web interface shows that the line is registered, but the GTI control panel indicates that it is not registered. This device supports multiple provisioning files in a hierarchy. The "Phone Number" field may contain only digits, and *must* contain at least one digit, or the base unit won't try to register it. Within those restrictions it can be any number you want. When using the handset (or base unit) to select a line from which to dial, this number will be displayed

61 next to the entry for the associated line. I think Panasonic's idea is that this is supposed to be a DID associated with the SIP address, but it doesn't have to be. Enter your SIP username in the "Line ID" field and again in the "Authentication ID" field. If you are registering a GTI main account your SIP username will be your User account number. If you are registering a sub-account it will be your main account number followed by an underscore followed by the sub-account number. You can register at any GTI server (e.g. "sip.ncsvoice.com ") but you must enter your chosen server name in the "Registrar Server Address" and "Proxy Server Address" and "Service Domain" fields. After saving these settings, check the "VoIP Status" page in the "Status" section to see if the status of this line is "registered". This product can register up to eight SIP addresses at once. If any of the SIP addresses you configured into the device do not show their status on this page as being "registered", then the status light on the base unit will flash yellow. (This is not mentioned in either the user guide nor the admin guide for this product.) The above information applies to both the TGP-550 (where the base unit has a handset, dial pad, LCD display, etc.) and the TGP-500 (where the base unit is a faceless black box that sits in your network closet). Top Page Error! Bookmark not defined. Pirelli DP-L10

62 Pirelli DP-L10 Product: Pirelli DP-L10 Company: Pirelli Overview: Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet. The Dual Phone enables mobile phone features such as Internet browsing, , SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings. See Configuration Details Configuration Details 1. Go into the WLAN/SIP settings phone's menu 2. Complete the following fields: Username: (your GTI username) Password: ******* (account password) Auth. Name: (if required) Domain name: sip.ncsvoice.com (one of our multiple servers) Local port: 5060 (default SIP Port) Proxy server: sip.ncsvoice.com Proxy port: 5060 Register server: sip.ncsvoice.com Register port: 5060 Register period: 3600 Outbound server: sip.ncsvoice.com (if required) Outbound port: 5060 RTP audio Port: (use default value) RTP pkt.period: (use default value) Preferred codec: G711u Top Page Error! Bookmark not defined.

63 Polycom SoundStation IP 4000 Conference Phone Polycom SoundStation IP 4000 Conference Phone Product: Polycom SoundStation IP 4000 Conference Phone Company: Polycom Overview: the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the benefits and versatility of a SIP enabled business. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated. The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference. Configuration Details Get the IP Start the Phone and it will get an IP from the DHCP Server, then it will display the IP on the Screen. Configure Settings Go to Lines tab and then to Line 1 Change Settings according to your Account values

64 Click the Submit button to save your settings. Go to General tab and then to Audio Processing Set your Codec to the following:

65 Click the Submit button to save your settings. Top Page Error! Bookmark not defined. Polycom SoundPoint IP 501, 550, 650, etc. Polycom SoundPoint IP 501 Product: Polycom SoundPoint IP 501, 550, 650, etc. Company: Polycom Overview: The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors IP solutions. As protocols develop and standards evolve, it s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment. Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs

66 Configuration Details Get the IP Start the Phone and it will get an IP from the DHCP Server, then it will display the IP on the Screen. Configure Settings Enter the IP address on your web browser (PC and Device should be on the same network) The browser should load the configuration page for the IP 501 Click on Sip Link Enter User and Password (default values are user: Polycom, pass: 456) Enter the Following values: Click on Submit. This will save settings and reboot unit. After the Device has rebooted, click now on Line Link and set your Identification values:

67 Click on Submit. This will save settings and reboot unit. After the Device has rebooted, your Device will be ready to be used with GTI Top Page Error! Bookmark not defined.

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