Recommended equipment

cisco-spa112

Cisco SPA112 is a 2-port adapter providing advantages of VoIP without the need to upgrade existing analog phones. You can use your standard landline phone in order to make calls via the Internet (VoIP) while receiving better voice quality. You can also save phone numbers and send faxes. Cisco SPA112 has a compact design and meets international standards of voice and data transmission. You can use it at home and in office.

SPA8000

Cisco SPA 8000 is an 8-port VoIP adapter which allows to connect up to 8 analog phones. It has a compact design and meets international standards of voice and data transmission. You can use it at home and in office.

CiscoSPA901

Cisco SPA901

Cisco SPA901 is perfectly compatible with IP PBX. In case you need to equip your office with a lot of phones, this model will help you cut expenses without compromising the quality. It’s also very convenient to use this model in warehouses and other locations where it’s easier to attach the phone to the wall rather than put it on the table.

CiscoSPA921

Cisco SPA921

Cisco SPA921 is a multifunctional and easy-to-use IP-phone. It is suitable for use at home, in organizations as well as at medium-size enterprises, in IP-PBX environment and large IP-centres.

SPA921 is equipped with graphic display, speakerphone and 2.5 mm headset output. With the help of this device you can simultaneously receive two calls on the same line. It also has such functions as: 3-way communication, transfer of call with notification. It also allows to put on hold the ongoing conversation in order to receive an incoming call. You can also assign to the line a unique number or make it a general one assigned to several devices.

CiscoSPA501

 Cisco SPA501

  • IP-phone Cisco SPA501G – is a handy and affordable 8-line IP-phone which is most widely used at reception lobbies and in waiting rooms. This model is not equipped with a display, but has removable paper plate for recording speed dial numbers.
  • Cisco SPA501G has two switch outputs Ethernet supporting IEEE 3af compatible with PoE (Power Over Ethernet). IP-phone Cisco SPA501G also has an indicator of pending messages, 4 navigation and 12 dialing buttons; illuminated buttons of volume adjustment, menu setup and voicemail.
  • SPA501G is a perfect model to use in organizations which require nothing more than a simple multichannel phone supporting PoE. SPA501G supports the standard protocol SIP 2.0 as well as its own protocol Cisco SPCP (Smart Phone Control Protocol).
CiscoSPA962

Cisco SPA962

Attractive design and a broad range of functions make Cisco SPA962 a universal model to use in the rapidly changing environment of IP-telephony, IP-PBX as well as in big call-centers. This model brings VoIP technologies to the next level at the same time remaining highly affordable and easily serviceable.

The Cisco SPA962 is equipped with 6 fully functional channels (dual switched Ethernet ports). It also supports 802.3 powered by the switch, has a high-resolution color display, loudspeaker, and 2.5mm headset output. All channels of the phone may be assigned one single phone number or individual numbers.  The model is not equipped with an ordinary power adapter and is powered only through PoE. This phone is supplied with PA100 adapter.Каналы телефонного аппарата можно настраивать как на один номер, так и каждый канал на свой номер. В комплектацию телефона Cisco SPA962 не входит блок питания. Внимание. Телефон Linksys SPA962 работает только через PoE*. Для телефона этой модели необходим адаптер PA100 (поставляется в комплекте).

CiscoSPA932

Cisco SPA932

Cisco SPA932 – is a 32-button complementary console for Lynksys SPA962. Key features:

  • 3-color indicator
  • May be connected to another console and thus you will have 64 buttons instead of 32
  • Support of Broadsoft
  • Support of Busy Lamp Field
  • Powered by SPA962 – no need to use additional power adapters

Support of Broadsoft Busy Lamp Field

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Setup manuals

3CX Phone

1. Download 3CXPhone here and then install the program. We recommend installing the version 6.2. When the installation is completed, run 3CXPhone and click on «Create Profile» to create an account.

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2. Press «New» in order to create a new account.

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3. In account settings fill in the following information:

Account name: for example Stream Telecom

Caller ID: the name which will appear during incoming calls

Extension (internal number): for example 101

ID: your sip-number (for example 80011000123)

Password: Your sip-number password

External IP: sip.streamtelecom.eu

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4. If settings are adjusted correctly you will see a status «On Hook» (available)

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X-Lite

  1. Download X-Lite here
  2. Open the menu «Softphone – Account settings»
  3. Enter the following information:

Account name: Stream

User ID: Your sip-number (for example 80011000123)

Domain: sip.streamtelecom.eu

Password: your password

Display name: your extension or name (for example 123)

Authorization name: your sip-number (for example 80011000123)

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4. After settings are saved the program status will change to «Available».
Now you can make calls.

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PortGo

  1. Download PortGo for Windows here.
  2. Select «Add account»
  3. Enter the following information:

Username: your SIP-number (For example 80011000123)

Password: your password

Server: sip.streamtelecom.eu

Automatic authorization: check the box and press «Enter»

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3. The program is ready for use after its status changed to «Registered».

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ZoIPer

1.Download ZoiPer here.

2. In the account master select «SIP» as an account type.

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3. Create new account.

user/user@host: 80011000123 — где 80011000123 Ваш sip-номер

Password: Ваш пароль от sip-номера

Domain/Outbound proxy: sip.streamtelecom.eu

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4. Account name: Stream

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5. The account is added, press «Close»

6. Now you can make calls.

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VentaFax

Download VentaFax here.

This program is suitable for receipt/sending of faxes via Internet using a laptop or computer.

Program setup

1. In the menu «Connection» select «use internet telephony»

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В меню «IP телефония» жмем «Добавить»

 

2. Enter the following information:

Name: stream

SIP-server: sip.steatelecom.eu

SIP ID or username: Your SIP number (for example 80011000123)

Password: your password

Authorization name: your SIP number.

Outgoing proxy-server: sip.streamtelecom.eu

Forbid Protocol T.38 during receipt and transmission of data (check the box)  

Way of DTMF-signals transmission: RFC2833 coding

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Press «OK» and the account status will change to «Registered»

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3. The program is ready for use; now you can now send faxes.

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X-Lite

  1. Download X-Lite here
  2. Open the menu «Softphone – Account settings»
  3. Enter the following information:

Account name: Stream

User ID: Your sip-number (for example 80011000123)

Domain: sip.streamtelecom.eu

Password: your password

Display name: your extension or name (for example 123)

Authorization name: your sip-number (for example 80011000123)

 

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4. After settings are saved the program status will change to «Available».
Now you can make calls.

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Telephone

1. Download Telephone here

2. Select the menu «Phone-Settings»

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3. Enter the account information from your SIP-account provider:

Name: Stream

Domen: sip.streamtelecom.eu

User: Your SIP number (for example 80011000123)

Password: Your password

4. When the account is added the program status will be changed to «Available». Now you can make calls.

ZoIper

1. Download Zoiper here.

2. Run Zoiper and select «Settings».

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3. Select the entity «Accounts».

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4. Click on «+» to add an account.

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5. Select «SIP Accounts».

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6.

Here you have to fill in all the sections:

Account name: your number or name (for example 80011000123)

Domain: sip.streamtelecom.eu

Username: your sip-number (for example 80011000123)

Password: Your sip-number password

Click on «Register»

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7. The program is ready for use. Now you can make calls.

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Bria

1. Download Bria here (fee-based SIP-client).

2. Run Bria and click on «Settings».

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3. Select the entity «Accounts».

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4. Click on «SIP» to add account.

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5. Here you have to fill in all the sections:

Account name: your number or name (for example 80011000123)

Domain: sip.streamtelecom.eu

Username: your sip-number (for example 80011000123)

Password: Your sip-number password

Press «On»

6. Congratulations. Now you can make calls.

Linphone

1. Download Linphone here.

2. Run Linphone and press «Next».

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3. Click on «Enter SIP server».

4. Fill in the following information:

Domain: sip.streamtelecom.eu

Username: your sip-number (for example 80011000123)

Password: Your sip-number password

5. Click on «Apply». If settings were entered correctly, the connection staus will change to «Registered».

 

6. Now you can make calls.

Linphone

1. Download Linphone from GooglePlay.

2. Run Linphone and click on «Next».

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3. Press «I already have registered SIP account».

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4. Now enter the registration data:

Domain: sip.streamtelecom.eu

Username: your sip-number (for example 80011000123)

Password: Your sip-number password

5. If the infromation was entered correctly press «Apply». The connection status will change to «Registered».

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6. In the «Settings» entity you can adjust additional functions.

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7. In the entity «Network» you can adjust the application to work only through Wi-Fi connection

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8. In «Advanced» entity you can set autorun of Linphone.

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ZoIper

1. Instal Zoiper from GooglePlay.

2. After installation of Zoiper select «Settings».

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3. Select «Accounts»

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4. Press «Add account»

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5. Press «Yes»

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6. Select «Manual Configuration»

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7. Select SIP

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8. Enter the following information:

Account name: Stream

Host: sip.streamtelecom.eu

Username: Your sip номер (например 80011000123)

Password: Your password

Press «Save».

9. The account is adjusted and the service is ready for use.

Linphone

1. Install Linphone.

2. Run Linphone and click on settings («gear-wheel» button).

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3. Enter registration data:

Username: sip number (for example 80011000123)

Password: password

Domain: sip.streamtelecom.eu

Proxy: sip.streamtelecom.eu

Press «Save»

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4. The status will be changed to «Registered».

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5. For adjustment of audio codeсs access «Settings» and select «Audio-settings – Codecs Options»

Compatible codecs: G711.a(alaw), G711.u(ulaw), G729, gsm, speex

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Gigaset

1. Enter the web-interface of the phone. You can make it via any web-browser using the IP-address of the database.

2. Open the entity «Telephony» «Connections». Check the «Active» box near the account which you want to adjust and press «Edit»

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3. Enter the following registration credentials:

Name or connection number: stream

Access password: your SIP number password

Username: your SIP login (For example 80011000123).

Display name: your SIP number (For example 80011000123).

Domain: sip.streamtelecom.eu

Proxy-server address: sip.streamtelecom.eu

Registration server: sip.streamtele.com

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4. Setup of calls routing.

Open the «Function of the number» in the «Telephony» entity. Here you can set up the default route for outgoing calls (via VoIP or a landline)

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5. The adjustment of RTP-ports 10000-20000:

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D-Link DPH-150S

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In order to access the web-interface of the phone you’ll need to use its IP address which you can find in the Menu.

After that open the «Voice» module, select the «SIP» entity and enter the following information:

Server Address: sip.streamtelecom.eu

Authentication User: your SIP login (For example 80011000123)

Authentication Password: your SIP number password.

SIP User: your SIP login (For example 80011000123)

Display Name: your SIP login (For example 80011000123) or name

Domain Realm: sip.streamtelecom.eu

Server Name: sip.streamtelecom.eu

Registration Expires: 300

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To save settings click on «Apply»

Linksys SPA921

Access the web-interface of the phone via any web-browser using the IP address of the device. You can find the IP-address by clicking on menu (piece of paper button), then «Network»«Current IP».

In the upper left corner of the page click on «Advanced» mode.

Open the «Ext1» entity and enter the following credentials:

Proxy: sip.streamtelecom.eu

Display Name: your SIP login (For example 80011000123)

UserID: your SIP login (For example 80011000123)

Password: your password

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In order to save settings press «Submit All Changes»

Linksys SPA921

Войдите в веб-интерфейс телефона. Для входа в веб-интерфейс Вам нужен IP адрес базы и зайти на него с любого браузера. Узнать текущий IP-адрес телефона можно нажав кнопку меню (Листок) затем Network => Current IP.

В правом верхнем углу страницы перейдите в режим Advanced

1. Откройте вкладку Ext1 и укажите следующие данные:

Proxy: sip.streamtelecom.eu

Display Name: Ваш SIP номер (Например 80011000123)

UserID: Ваш SIP номер (Например 80011000123)

Password: Ваш пароль

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Для сохранения настроек нажмите Submit All Changes

Cisco SPA112

The Cisco SPA 112 model is the best choice for the companies with existing network infrastructure.

Outputs, indicators and their functions:

Rear panel:

Outputs of the phone adapter are located on the rear panel.

  1. POWER – output for connection of the power adapter.
  2. INTERNET – output for the connection of landline phones and network (Rj-11)

Front panel:

The front panel contains four indicators reflecting the battery charge level, status of the Internet connection as well as of two phone lines.

  1. POWER – is on when the phone adapter is on and connected to the Internet. When there is no connection the indicator blinks.
  2. INTERNET – is switched on when there is internet connection. Blinks when there is a traffic flow through the port.
  3. PHONE 1/2 – is switched on when phones (faxes) are connected. Blinks during conversation or picked-up phone.

Connection of the cables to the phone adapter:

  1. Connect the analogue phone or fax to the adapter via PHONE 1 using the phone cable Rj-11.
  2. Repeat the previous step for the output PHONE2.
  3. Connect your computer to the ETHERNET output using the Rj-45 cable.
  4. Connect your ETHERNET cable to the INTERNET port using the Rj-45 cable.
  5. Connect the power adapter to the POWER output.
  6. Wait until the POWER indicator will indicate that the phone adapter is switched on.
    7. Switch on the computer. Use the Ethernet port of Cisco SPA112 to connect to the router or multiplexer installed in office or at home. This model supports only two types of connection: static and dynamic IP. Apart from Ethernet outputs there are two phone outputs Rj-11 on the rear panel of the adapter. These are FXS outputs for connection of the customer’s equipment. These outputs are to be used for connection of telephones, faxes or outgoing lines of the exchange.

Connection:

Connect the dedicated Internet-line (the cable provided by your Internet-provider) to the «Internet» output of the adapter. Connect the phone to the adapter via the output «Phone 1». As a rule the equipment has default settings which may be incompatible with your network. If the equipment received from the provider already has default settings, try to connect it.

The most widespread connection to the provider’s network is DHCP mode (as a rule set as default). You can check whether the equipment is operational by following these steps:

1. Connect the device to the provider’s network (Internet output) INTERNET (WAN) indicator will switch on.
2. Connect the phone to the adapter (output Phone 1)
3. Put the phone device to the tone mode
4. Dial the number; you will hear the ring-back tone.
Required settings for connection of the adapter (gateway): Cisco SPA112
1. Connection type (DHCP, StaticIP,…)
2. For the StaticIP — ….
3. Authentication via MAC address
In case you are sure that the device was not previously adjusted according to your needs, it is recommended to return back to default settings. Though be aware, that there is no reset switch in the device. The only available solution is to change the network settings via the menu using the keyboard of the device.

Most frequently codes of voice menu
Access to the voice menu : ****
Verification of the connection method: 100
Identification of the connection method: 101-1 DHCP; 2 Static IP (may vary depending on the model; check using the command 100; in SPA 112 connected as enable/disable in relation to DHCP)
Verification of WAN IP address: 110
Identification of statistic WAN IP address: 111
Verification of WAN mask: 120
Identification of WAN mask: 121
Verification of WAN gateway: 130
Switching on/off the web-server:7932-1 switching on; 2 switching off;
Verification of LAN IP address: 210
Return to default settings: 73738
In case of absence of external DHCP server it’s necessary to install a static IP address,  connect devices to the computer through the network cable (RG45), run web-browser and connect to the device by entering the IP address of the device in the address-bar of the web-browser.

External WAN Internet connection:

In order to adjust your adapter you have to know the parameters of your network. You can find them by following simple steps. In the low right corner of the «Desktop» click on network connections entity, and stick to the instructions provided on the images below:

Now you know the parameters for the setup of the adapter.
Set up of the adapter Linksys SPA 2102 using the web-interface:

Run an internet browser on your computer:
Type the following info in the address bar of the web-browser: 192.168.0.1/admin/advanced and press «Enter».
In the window «Router-Status» select the entity «WAN Setup»:
Choose the «Static IP» option in the pop-up menu «Connection Type».
In the popped-up entity enter the information related to your «network connection»:
Static IP (IP address): XXX.XX.X.XXX – considering changing 3 last digits (0-225) .
Gateway (main gateway): XXX.XX.X.X
NetMask: 255.255.255.0
DNS
– select any you like, or use recommended ones: pr: 4.2.2.2. sdr.: 208.67.220.220.
Click on the button: Submit All Changes

Connection and setup of adapter Cisco SPA112:

Apart from two main types of connection (static and dynamic IP) which may be used for connection of Lynksys SPA2102 to the existing network, this model also supports three different types of PPPoE connection working in static and dynamic modes of IP-address assignment. On this page you can set up the type of provider’s connection. After you’ve selected the connection type windows which are important for the provider will pop-up.

After you’ve selected the connection type the form which you have to fill up in order to set up this particular connection will pop-up. You have to fill in all the required sections of the form.
LAN connection:
After that you have to set up an internal LAN connection. In order to access relevant settings please select the entity «Admin Login» and then click on «Advanced» in the right upper corner of the page.

Here you can also select the network service type (NAT or Bridge), indicate the address of LAN section, turn on/off DHCP server, set up the time of saving of dynamic IPs and their number as well as to fix the IP address for 10 subscribers.

Voice settings:
SIP entity
like two other entities «Provisioning» and «Regional» has special options which allow to conduct a precise adjustment of the adapter for its use under different circumstances. Cisco SPA112 supports two independent VoIP channels, which can be set up via Line1 and Line2 entities.
1. Select the entity «Voice» in the settings window
2. Then select «Line 1» entity
3. Fill in all required fields in «Line 1» entity with information provided by the Stream Telecom manager, for example:

SIP Port:  5060
Proxy: sip.streamtelecom.eu
User ID: You sip number (username). For example — 80044ххх ххх
Use Auth ID: yes
Auth ID: You sip number (username). For example — 80044ххх ххх
Password: your sip password
Register Expires: 3600
Make Call Without Reg: yes
Preferred Codec: g.711, g.729a

You can use default settings for other sections.
In order to save your changes click on «Submit All Changes» in all entities.
If you do everything correctly you’ll hear a «dial tone», while in the «Voice-Info» entity you will see the following: Registration State: Registered.

It means that the device is properly adjusted and is ready for use.

Yealink

1. Enter the web-interface of the phone and open the entity «Account»
Enter the following information: :

Access password: your SIP number password

Username: your SIP login (For example 80011000123).

Display name: your SIP number (For example 80011000123).

Domain: sip.streamtelecom.eu

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Grandstream DP-710 / 715

1. Enter the web-interface of the phone and open the entity PROFILE 1:
Enter the following data::

Primery SIP Server: sip.streamtelecom.eu

Outbound Proxy: sip.streamtelecom.eu

Register Expiration: 10

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2. Open the entity HANDSETS:

SIP User ID: Your SIP number (for example 80011000123)

Authenticate ID: Your SIP number (for example 80011000123)

Authenticate Password: Your password

Name: Your SIP number (for example 80011000123)

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Escene

1. Enter the IP-address of the device in the address bar of the web-browser in order to access the web-interface of the phone.

You can find out the current IP-address of the device by clicking «С», the second line of the menu – IP.
In the menu column select the entity SIP accounts and then select «Account 1» and enter the following information:

Username: your SIP login (For example 80011000123).

Display name: your SIP number (For example 80011000123).

Access password: your SIP number password

Domain: sip.streamtelecom.eu

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Frequently asked questions

Echo during conversation

Q:  When I make a call via X-lite I hear my own voice. What can be done?

A:  Very often this problem is faced by owners of the computers with built-in sound card. This effect is caused because stereo mixing of incoming sounds is switched on by default. We’ll show how to address this issue using the Realtek HD Audio: 1 program. Open the Control Panel (Start — Control Panel):

2. Click on «Realtek HD audio configurations»

3. Switch off the function «Stereo Mix» in the «Mixer» entity of «Record» module.

In case you have a built-in sound card follow the same logic: access the sound settings, then click on «Mixer» and turn off the function «Stereo Mix».

Disruption of speech

Q: Why during conversations the speech is occasionally or routinely disrupted?.
A: Such problem, as a rule, occurs when the quality of the Internet connection is poor. Check the Internet channel for any malfunctions using ping as well as try to monitor the lags. They should not vary significantly.

ТYou also have to pay attention to the speed of internet connection. For proper functioning of equipment, the speed of your internet connection should be not less than 100 Kbps per 1 phone line (under g.711a codec).

One-way audibility

Q: I can’t hear the interlocutor during the conversation while he/she clearly hears me.
A: This problem may arise because of settings of your local network. Most probably you are using a NAT router. Check its settings.

Conversation interrupts

Q: Why conversation interrupts sometimes during a phone call?

A: One of the most plausible reasons – loss of Packets during transmission of data via the Internet network. You can trace the transmission of ICMP Packets to the IP-address of the subscriber by typing the command «tracert» in the command line of the Windows, WinMTR, PingPoitter. Or you can approach your Internet provider for assistance.

Standard settings for Firewalls

Your program phone or equipment (the SIP agent) may be connected to the Internet through the network screen. In this case in order to correctly adjust your SIP agent you have to adjust your network screen as follows:

  • allow incoming traffic under UDP protocol from addresses 136.243.23.46, 136.243.23.9, 136.243.23.8 to all ports;
  • allow outgoing connections and outgoing traffic under UDP protocol through addresses: 148.251.196.61, 136.243.23.9, 136.243.23.8 to the following ports:
  • 5060 — for signaling information of SIP protocol;
  • From 8000 to 20000 for voice traffic (RTP).

Standard settings for SIP equipment

Login – your SIP-ID
Password – your SIP-ID password
Registration server – sip.streamtelecom.eu
Registration port – 5060
Outbound server status – on
Outbound server – 5060
STUN-server status – off
STUN-server – N/A
Codec – g.711a (PCMA), g729a, GSM
Method of DTMF transfer – rfc2833
Fax transfer – inband in codec g.711a/T.38

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