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

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

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 PAP2 .......................................................................................................................... 11

Configuration Details ................................................................................................................... 11

Configuration Screens ................................................................................................................. 13

How to avoid the long delay to hear the ringtone ...................................................................... 15 Cisco Linksys PAP2T ........................................................................................................................ 17


Configuration Screens .............................................................................................................. 19

Customer Submitted Information ............................................................................................. 21

How to avoid the long delay to hear the ringtone ......................................................................... 22

Linksys PAP2T ATA Adapter Reset Procedure ................................................................................ 23 Cisco SPA112 .................................................................................................................................. 24

Configuration Details ............................................................................................................... 24 Cisco Linksys SPA942 NA ................................................................................................................. 27

Configuration Details ............................................................................................................... 27

See also ...................................................................................................................................... 29

What is a Dial Plan? ................................................................................................................. 29

Digit Sequence ........................................................................................................................ 29

Examples ..................................................................................................................................... 30

See also ...................................................................................................................................... 31

References ............................................................................................................................... 31 Cisco IP Phone 7940/7960 ............................................................................................................... 31

Configuration Details ................................................................................................................... 32 Cisco SPA2100 Phone Adapter ......................................................................................................... 33


How to avoid the long delay to hear the ringtone ...................................................................... 35 Cisco SPA2102 Phone Adapter with Router ...................................................................................... 36


How to avoid the long delay to hear the ringtone ...................................................................... 38

Dial Plan for Linksys ATAs ......................................................................................................... 39

Dial Plan ................................................................................................................................. 39

Digit Sequence ........................................................................................................................ 40

Examples ..................................................................................................................................... 41

See also ...................................................................................................................................... 41

References .................................................................................................................................. 41 Cisco SPA504G Phone ..................................................................................................................... 42

Configuration Details ................................................................................................................... 42 Grandstream HandyTone 286 .......................................................................................................... 44

Configuration Detail ................................................................................................................ 45

Message Waiting Indicator ....................................................................................................... 45 Grandstream HandyTone 486 .......................................................................................................... 46

Configuration Detail .................................................................................................................... 46

Grandstream HandyTone 502 ...................................................................................................... 47

Configuration Detail .................................................................................................................... 48

Mediatrix 4100 Series .................................................................................................................. 48


Optional Setting: ..................................................................................................................... 52 Netgear WGR615V .......................................................................................................................... 54

Configuration Detail .................................................................................................................... 54 OBi110 ............................................................................................................................................ 56

Configuration details ................................................................................................................... 56

Default Dial Plan: ..................................................................................................................... 57 Panasonic KX-TGP 550..................................................................................................................... 59

Configuration Details ................................................................................................................... 60 Pirelli DP-L10 .................................................................................................................................. 61

Configuration Details ................................................................................................................... 62 Polycom SoundStation IP 4000 Conference Phone ............................................................................ 63


Configure Settings ................................................................................................................... 63 Polycom SoundPoint IP 501, 550, 650, etc. ....................................................................................... 65


Configure Settings ................................................................................................................... 66 Siemens Gigaset C450-Ip ................................................................................................................. 68

Configuration Details ................................................................................................................... 68 SNOM 320 VoIP Phone .................................................................................................................... 70

See Configuration Details ............................................................................................................ 71

Using the "Retrieve" Voice mail Key ............................................................................................. 72 Snom m3 VoIP Phone ...................................................................................................................... 73

Configuration Details ................................................................................................................... 73 Telco AC-211 .................................................................................................................................. 74

Configuration Details ................................................................................................................... 75 Yealink SIP-T28P (VSRF) ................................................................................................................... 76

Configuration Details ................................................................................................................... 77 Zycoo ZP502 ................................................................................................................................... 79


BASIC SETTING ........................................................................................................................ 81 Polycom IP450 Setup & User Guide ........................................................................................... 83

SoundPoint® IP 450 HD Quick User Guide V002 ................................................................... 83

1.1. Key Location .................................................................................................................. 83

1.15. Directories ................................................................................................................... 84

1.16. More Detailed Information from the Manufacturer ...................................................... 84 Manually configuring a Polycom SoundPoint IP 320, 321, 330, 331, 450, 550, 560, 650, 670 and SoundStation IP 6000, 7000 with 3CX Phone System ............................................................... 84 Polycom SoundPoint® IP 335 HD Voice Quick User Guide .................................................................. 87

KEY LOCATIONS ................................................................................................................. 87 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 ..................................................... 88

Set up instructions ...................................................................................................................... 89 Cisco SPA301-G1 User Guide .................................................................................................... 90

Phone Layout ........................................................................................................................... 90

Setup detail ................................................................................................................................ 90 Cisco SPA303G User Guide ....................................................................................................... 91

Phone Layout ........................................................................................................................... 91

Setup detail ................................................................................................................................ 91

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

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 [ ] to confirm selection

Select IP&MAC Addresses and press [ ]

The IP Address will appear on the screen.

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.

Screen Name: John Smith (value of your choice) Phone Number: 123456 (Your GTI username) Caller ID: 123456 (Your GTI username) Authentication Name: 123456 (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)

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

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.


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: 123456 (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

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

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.


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: 192.168.1.2)

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 http://192.168.1.2) Replace 192.168.1.2 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: 123456 (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]11|822|0|00|[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: 0.020

Configuration Screens PAP2 Configuration Screen

PAP2 Configuration Screen

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. 5554441234#). 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).

http://www.toao.net/25-linksys-ata-configuration

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.

___________________________________________________________________________

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, voicemail, 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

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).


Step 1




The system should now playback the IP Address your device has been assigned. (Example: 192.168.1.2) Step 2




Under the LINE 1 Tab, Find the following fields and fill them with the following information Nat Keep Alive: 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: 123456 (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO







Configuration Screens

PAP2T Configuration Screen PAP2T Configuration Screen

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.

http://www.toao.net/25-linksys-ata-configuration

http://www.cisco.com/en/US/products/ps10029/index.html




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. 5554441234#).

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).

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.

73738#

(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.


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 http://www.cisco.com/go/smallbizvoicegateways. 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 1 1 0 # slowly. The current IP address will be read back.

If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address

http://www.cisco.com/go/smallbizvoicegateways

http://www.cisco.com/go/smallbizvoicegateways

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 10.1.27.2). 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, http://10.1.27.1). 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]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

(note: replace 555 in the dial plan with your area code)

Click Submit to save settings.

Step 4

https://supportforums.cisco.com/docs/DOC-9900

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

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.


Step 1





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






Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: 123456 (Replace with your GTI username) Password: ******** (Type in the account password) Use DNS SRV: NO







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 Function

0 1 2 3 4 5 6 7 8 9 * #

x

[sequence]

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

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 *.

You can use a period to accept zero or more entries of a give digit.

. (period) 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.

<dialed:substituted> 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 6782345, the system transmits 15556782345.

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.

This override the setting in the Short inter-digit timer or Long inter-digit S0 or L0

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)

<: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

Cisco SPA2100 Phone Adapter Cisco SPA2102 Phone Adapter with Router Cisco Linksys PAP2 Cisco Linksys PAP2T

IP Phones

Cisco SPA300/500-series 'phones Cisco Linksys SPA942 NA

References

Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue telephone adapters

Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones


Cisco IP Phone 7940/7960

http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf


http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf

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.


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. 123456)

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.


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.


Step 1




The system should now playback the IP Address your device has been assigned. (Example: 192.168.1.2) Step 2






Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: 123456 (Replace with your GTI username) Password: ******** (Type in the account password)

Use DNS SRV: NO


Step 5 (Optional)




(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

Step 6



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:


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).

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.


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://192.168.0.1/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


click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".

Step 3



Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: 123456 (Replace with your GTI username) Password: ******** (Type in the account password)

Use DNS SRV: NO


Step 4 (Optional)





Step 5



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:


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.

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 5081.

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:


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).

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

0 1 2 3 4 5 6 7 8 9 * # You can use any of these characters to represent a key pressed in your

keypad.

x This represent any character on the phone.

You can enter characters between brackets to create a list of accepted

digits.

[sequence] 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.

This is used for sequence substitution, you can use this to indicate that

certain numbers dialed are replaced by other characters. The dialed digits

<dialed:substituted> 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 6782345, the system transmits 15556782345.

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.

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

Cisco SPA300/500-series 'phones Cisco Linksys SPA942 NA

References

Cisco - Administration Guide: SPA2102, SPA3102, SPA8000, SPA8800, PAP2T analogue

telephone adapters Cisco - Administration Guide: Cisco SPA300/SPA500 series and Cisco Wireless-G IP phones




http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa500_admin.pdf

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.


Step 1





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






Proxy: sip.ncsvoice.com Register Expires: 180 Proxy Fallback Intvl: 180 Display Name: John Smith (Replace with your name or company name) User ID: 123456 (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)



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:


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


Grandstream HandyTone 286


Product: Grandstream HandyTone 286

Company: Grandstream

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: 123456 (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 1-877 number] )

Message Waiting Indicator

To enable this feature with your Grandstream HandyTone 286 you need to associate a voicemail with the account (or subaccount) you have registered in the device and set the SUBSCRIBE for MWI option to YES


Grandstream HandyTone 486 Grandstream HandyTone 486

Product: Grandstream HandyTone 486


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.


Step 1:


Step 2:


SIP Server: sip.ncsvoice.com (one of our multiple servers) SIP User ID: 123456 (Your GTI username) Authenticate Password: ********* (Account Password) User ID is phone number: No SIP Registration: Yes NAT transversal : 10



Grandstream HandyTone 502 - HT502

Product: Grandstream HandyTone 502 - HT502


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


Step 1:


Step 2:


SIP Server: sip.ncsvoice.com (one of our multiple servers) SIP User ID: 123456 (your GTI username) Authenticate Password: ********* (account password) User ID is phone number: No SIP Registration: Yes NAT transversal (STUN): No


Mediatrix 4100 Series

Mediatrix 4102 Product: : Mediatrix 4102S and 4100 Series running DGW 2.0 firmware

Company: Media5 Corporation

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.


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: 192.168.1.2)

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: http://192.168.1.2) Replace 192.168.1.2 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).

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: 123456 (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

Step 5

Click the SIP Menu

Sub-menu Authentication

Click Edit on the first row

Apply to: Endpoint

Endpoint: Choose the FXS port for which you want to configure the Authentication

Validate Realm: disable

Username: 1234560 (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

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


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 (802.11 G) router that happens to have a built-in ATA. It is possible to use just the ATA part.


Go to http://192.168.61.1/set_voip.htm

Complete the following fields:

SIP Proxy: Your GTI server which is sip.ncsvoice.com

SIP Control Port: 5060 or 5080

...

Line Enable: Yes

Display Name: whatever you want call display to show

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.


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.

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: 123456 (Your VoIP.ms username)

Voice Services >> SP Service

AuthUserName: 123456 (Your GTI username) AuthPassword: ********* (Account Password)

Checking your Voicemail

If you encounter any issues when you try to check your GTI Voicemail. 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.

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

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.


Panasonic KX-TGP 550

Panasonic KX-TGP 550

Product: Panasonic KX-TGP 550

Company: Panasonic

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


Complete the following fields:

Phone number ID: 123456 (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: 123456 (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

http://www.voicesonic.com/panasonic/manuals/panasonic-others/TGP500-550-Admin-Guide.pdf

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

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 802.11b/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, e-mail, 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


1. Go into the WLAN/SIP settings phone's menu

2. Complete the following fields:

Username: 123456 (your GTI username) Password: ******* (account password) Auth. Name: 123456 (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


http://wiki.voip.ms/article/Pirelli_DP-L10

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.


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

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

Click the Submit button to save your settings.


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


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:

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


Siemens Gigaset C450-Ip

Siemens Gigaset C450-Ip

Product: Siemens Gigaset C450-Ip

Company: Siemens

Overview: The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required. The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.


Web configuration

Select Telephony In the Web Configuration Page Select VoIP Enter the following values:

o Authentication Name: 123456 (your GTI username) o Authentication password: *********** (account password) o Confirm your password: ***********

o Username: 123456 o Domain: sip.ncsvoice.com (one of our multiple servers) o Proxy Server Address: sip.ncsvoice.com o Proxy Server Port: 5060 o Registrar Server: sip.ncsvoice.com o Registrar Server Port: 5060 o STUN enabled: NO o Outbound proxy mode: Never

Click on Set button to save your settings


SNOM 320 VoIP Phone

Snom 320 VoIP Phone

Product: Snom 320 VoIP Phone

Company: Snom

Overview: Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with built-in, full-duplex speakerphone and three-party conference bridging.

See Configuration Details

This article makes reference to the basic configuration for the SNOM 320 phone using a GTI account.

The initial configuration is very simple, you will need to plug the SNOM 320 phone to the internet source (modem, router), this will assign the phone with a IP address.

You can find the SNOM's IP Address by using the menu display on the phone.

Type the IP Address of the SNOM in your preferred Web Browser .

The default login details are: Username: admin, Password: admin.

The SNOM configuration page should appear. From the menu screen click on Setup -> Identity/Line 1.

Make sure to use your own credentials such as:

Username/SIP UserID (your GTI sip number, or the sub account that you wish to register) Password. Preferred Server (The sip.ncsvoice.com server of your choice)

Using the "Retrieve" Voice mail Key

In the 'Function Keys' page on the web-interface of your phone, select 'Speed Dial' as the type, in the 'Retrieve' row, 'Number' field, right next to the 'Type' field, enter *97

Also make sure your accounts/sub accounts does have the MWI active (Mailbox assigned)


Snom m3 VoIP Phone Snom m3 VoIP Phone

Product: Snom m3 VoIP Phone

Company: Snom

Overview: The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands free mode, calling line identification (CLI) by displaying name, number and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.


Web Management

Get the Device IP by Pressing the Volume Up key. Open your browser and go to http://<DeviceIP> Login to web management (Default User : admin, default Password : admin)

Account Settings

In the Left Menu select Identity 1 Fill in the following values:

o Server is local: No o Display Name: John Smith (your name)

o Account: 123456 (your GTI username) o

Password: ******** (account password) o Registrar: sip.ncsvoice.com (one of our multiple servers)

o Authentication Username: 123456 (your GTI username) o Server Port: 5060 (default SIP port)

Click on Save button to save your settings. You will be redirected to the Main Page Select Identity 1 again and click Reboot to apply settings

After a few seconds, both status indicators in the upper left corner of the display will turn yellow, indicating that you have successfully configured your Snom m3


Telco AC-211 Telco AC-211

Product: Telco AC-211

Company: Telco

Overview: Telco AC-211 is a SIP ATA supporting two lines and numerous configuration

options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now defunct Sun Rocket service. This device works well with GTI once configured.


Web Management

Enter the IP in the address bar of your browser for the unit Enter the password for the unit and click Authenticate

WAN Settings

Go to WAN tab Select Auto Configuration tab and uncheck "Enable Automatic Configuration"

SIP Settings

Go to SIP tab Primary Server: Enter the following.

o Enter the server name. Example: sip.ncsvoice.com o Port 5060 o Domain Name: sip.ncsvoice.com o Check "Send Registration Request" o Dial Plan: (>#|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|0|[489]11|*9[7-8]|*75xx) o Transport: UDP o User/Phone Num: Your SIP Username o CallerID Name: Enter the CID you want to send o Port*: 5060 o AEC On: On o Authentication User Name: Your SIP Username o Password: SIP Password o Click Save SIP Settings

CODECS

Go to CODECS tab Force preferred CODEC Line 1: Set your preferred CODEC, recommended is G711U

o The CODEC set must be enabled in your account: Account Settings -> Advanced Click Save CODEC Configuration

Miscellaneous

Go to Miscellaneous tab Enter an NTP server

o Free NTP Servers available o Click Save Settings

Reset

Click the Reset button to reboot the gateway

Notes

Under the SIP Server Settings the device supports a fail over server, but at this time the server does not support this. Leave this blank or you will have problems receiving incoming calls.

The dial plan supports 10 & 11 digit dialing, *97 & *98 for voice mail, *75 dialing from the phonebook, 411, 811, & 911.

When modifying the dialing plan >#| is required at the start of the plan, though I don't know why.


Yealink SIP-T28P (VSRF)

Yealink SIP-T28P

Product: Yealink SIP-T28P

Company: Yealink

Overview: Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is

equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.


Get the IP Address

Press the OK button The IP Address should appear on screen

Web Management

Open your web browser and go to http://<ipaddress> Login to web management (Default User : admin, default Password : admin)

Account Settings

Go to Account tab Chose one account

Fill in the following values: o Account Active: On

o Label: 123456 (your GTI username) o

Display Name: John Smith (your name)

o Register Name: 123456 (your GTI username) o User Name: 123456 (your GTI username) o Password: ********* (account password) o SIP Server: sip.ncsvoice.com (one of our multiple servers)

Click on Confirm button to save your settings.


Zycoo ZP502

Product: Zycoo ZP502

Company: Zycoo ZP502

Overview: ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broad soft, Cisco call manager etc.


Step 1

Once you confirm this phone is connected to your network, from your PC ( which should be connected to the network too ), enter the IP address of the wan port in this phone as the URL (e.g. http://xxx.xxx.xxx.xxx/ or http://xxx.xxx.xxx.xxx:xxxx/). If you do not know the IP address, you can look it up on the phone’s display by pressing navigation "down" button.

In the login page you will need to enter the username and password:

1. Default user with general level -> usernameguest / passwordguest

2. Default user with root level -> usernameadmin / passwordadmin

The default password of phone screen menu is 123. After you configure the

ip phone, you need click save button in config under Maintenance in the left catalog to save your configuration. Otherwise the phone

will lose your modification after power off and on.

Step 2

Once you enter on the configuration portal of this device, please go to VOIP section, there you can see a screen like this:

First you need to select the SIP line you will use, then you can set the following information on

BASIC SETTING

Server Address: sip.ncsvoice.com ( or IP address of any of our servers )

Server Port: 5060

Account Name: 1XXXXX or 1XXXXX_X ( here you need to enter your main account or a sub account )

Password : your account or sub account password

Phone Number: 1XXXXX or 1XXXXX_X ( here you need to enter your main account or a sub account )

Enable Register: Mark this box ADVANCED SIP SETTING

Register expire time: 180

NAT keep Alive interval: 180

Local Port: 5060 ( if this is blocked on your side, you can try with port 5080 )

NAT Keep Alive: Mark this box

Transport Protocol: UDP

Finally, you need to click on button "APPLY" to save these changes.

Step 3 (OPTIONAL)

In case you want to use IAX2 protocol, once you select IAX2 Tab you can see a screen like this:

There you can set the following information

IAX2 Server Addr: sip.ncsvoice.com ( or IP address of any of our servers )

IAX2 Server Port: 4569 ( if this is blocked on your side, you can try with port 5080 )

Account Name: 1XXXXX or 1XXXXX_X ( here you need to enter your main account or a sub account )

Account Password : your account or sub account password

Phone Number: 1XXXXX or 1XXXXX_X ( here you need to enter your main account or a sub account )

Local Port: 4569 ( if this is blocked on your side, you can try with port 5080 )

Enable Register: Mark this box

Finally, you need to click on button "APPLY" to save these

changes. Top Page Error! Bookmark not defined.

Polycom IP450 Setup & User Guide

SoundPoint® IP 450 HD Quick User Guide V002

1.1. Key Location

1.15. Directories

Press (Menu) button, the [features] soft key then the [Contact Directory]. Once inside pressthe arrow keys (V) or (^) and you will be able to browse through your extensions and address book entries. 1.16. More Detailed Information from the Manufacturer For detailed information, please consult the comprehensive SoundPoint IP 450 User Guide available at

www.polycom.com/support/voicedocumentation or

http://www.polycom.com/global/documents/support/user/products/voice/soundpoint_ip450_user_guide_sip3_1_englis

h_RevA.pdf

Polycom, the Polycom logo and SoundPoi

Manually configuring a Polycom SoundPoint IP 320, 321, 330, 331,

450, 550, 560, 650, 670 and SoundStation IP 6000, 7000 with 3CX

Phone System

Note 1: This guide has been tested with firmware version 3.3.1 and Bootrom 4.3.0. Be aware that different firmware revisions may have different web interface formats and functionality.

Note 2: This FAQ is based on a new phone OR one that has been reset to factory defaults. If in doubt, reset the phone to Factory Defaults.

The preferred method to configure a Polycom Phone is via Provisioning. You can however follow this guide to manually configure your phone.

http://www.polycom.com/support/voicedocumentation

http://www.polycom.com/support/voicedocumentation

http://www.phone.com/setup/polycom-ip450-setup-user-guide/%20http:/www.polycom.com/global/documents/support/user/products/voice/soundpoint_ip450_user_guide_sip3_1_english_RevA.pdf




Configuring the Phone to Register with 3CX Phone System

1. Start up the phone and identify its IP Address – using the menu key on the phone, go to the "Status" option,

then to the "Network" option, and select the "TCP/IP Parameters" option. For this example we will assume the IP Address of the phone is 10.0.0.108, and the IP Address of the 3CX Phone System machine is 10.0.0.11.

2. Point your browser to the web interface of the phone: http://10.0.0.108

3. We now need to set the phone to register with GTI . Click on the "Lines" tab, and configure as follows:

4. In the "Identification" section of the "Lines" tab:

a. Set the "Display Name" field to the name you want to appear on the Phone display. b. Set the "Address" field to the extension number you want to associate with this phone.

http://10.0.0.108/

c. Set the "Auth User ID" field to the, Please use GTI user ID d. Set the "Auth Password" GTI Authentication Password.

5. In the "Server 1" section of the "Lines" tab: Type sip.ncsvoice.com

a. Set the "Address" field to the IP Address of the GTI System machine – in this example, sip.ncsvoice.com

6. b. Set the "Port" field to 5060, which is the default - you can check the correct value from the "Settings -> Network"


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 Syste

Set up instructions

Select the extension you want to provision and go to the "Phone Provisioning" tab.

Identification 1- display name: enter the extension name 2- address: enter sip.ncsvoice.com ( on of GTI servers IP)

3- Auth user ID: enter the user id that was assigned to you from Smart Voice

4- Auth password: enter the password # that was assign to you by GTI network 5-

Label: enter your extension number

Server 1

1-Address: enter the ip address sip.ncsvoice.com ( one of GTI Server IP) 2-

Port: enter 5060 and follow the red arrow then click on save


Cisco SPA301-G1 User Guide

Phone Layout

1. Handset

2. Status Light and Message Waiting Indicator (MWI)

3. Voicemail button

4. Ringer on/off switch

5. Flash button

6. Redial button

7. Volume button

Setup detail

http://www.manualowl.com/m/Cisco/SPA509G/Manual/270068


___Top Page Error! Bookmark not defined.

_______________________________________________________________________

Cisco SPA303G User Guide

Phone Layout

1. Handset

2. Message Waiting Indicator (MWI)

3. Line buttons with status lights

4. Softkey (context sensitive)

5. Directional buttons

6. Voicemail, Menu and Hold buttons

7. Mute, Headset, Speaker and Volume buttons

Setup detail




Documents

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