Sippoint
FAQ
Using
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I wonder if I can talk to two users at the same time without switching between them so that they could hear each other, too. Basically, a conference call is a well known service.
And we provide this service. It has been tested with eyeBeam and Cisco 79x0.
eyeBeam:- establish connection for the first line;
- press the second line button and establish the second connection;
- press CONF button.
Cisco 79x0:
- establish connection;
- press Confrm button;
- establish the second connection;
- press Join button.
IP-АТС АГАT UX:
- establish connection;
- press FLASH button;
- establish the second connection;
- press FLASH button and them no later than in 1 second, 7 button
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How can I listen to auto answer messages?
Dial your own SIP ID number or *01 using your IP-telephone or software.
- To manage you mail box, press 1
- To save the message, press 9
- To return to the main menu, press #
- To record a hello message, press 2
- To record your hello message, press 1
- To save your hello message, press 5
- To return to the main menu, press #
- To start a conference call, press 61
- To remove any record in the conference call, press 65
- To exit the conference call, press #
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How can we write the telephone number you have provided us with to our clients? Will it include our city code and which one? Does it mean it will be a standard telephone number (495) 1234567 or a number without a specific city code?
We assign you a SIP ID instead of a telephone number. You can tell your friends/clients/partners your SIP ID as "sip:1234567@sipnet.ru" (if they can make IP calls). If you want to answer calls made from standard telephones, tell your friends/clients/partners your digital SIP ID, e.g. "1234567" or the city access telephone number. Applicable access telephone numbers are published on the City Access page in My Profile.
Also you can get a phone number in the Russian and Kazakhstanian codes. To clarify the terms of purchase, you can by email need@sipnet.ru
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What codecs does SIPNET support?
When you make phone calls from SIP ID to standard telephones, we support G.711(A and µ) and G.729. Any standard codec available to users can be used within SIP ID.
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What package routing protocols are supported by SIPNET?
Currently, SIPNET supports IPv6 routed network protocol. This version has higher mapping, integrated encoding capability and some other features.
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How to make calls
You can call your contacts in other locations:
- You will be able to place calls to SIPNET account holders as soon as you register with SIPNET.
- Upload at least $10 to your account and maintain your balance at this level to be able to make free calls to different destinations covered by the «Free Long-Distance and International Calls» promotion. To take advantage of the promotion, register your mobile number with SIPNET.
You can make calls:
- from your computer using the Sippoint softphone;
- from your computer using other Internet sofphones;
- from your mobile phone (iPhone, Android, Java) using the Sipnet application and other mobile phone applications;
- from your computer using the WebRTC service (not available in Trial Access mode);
- from IP phones as well as using adapters (VoIP gateways) for ordinary phones;
The simplest way to make calls involves installing the free Sippoint softphone on your computer. This type of connection lets you save on your entry costs, making it the most popular way of communication on the SIPNET network.
Number dialling rules
To place calls, enter the contact's number in the international format: country code - area code - contact number. Example: 4930XXXXXXX (Berlin), 7495XXXXXXX (Moscow), 396XXXXXXXXX (Rome)
Standard settings for softphones and VoIP phones
Configuring desktop and mobile softphones could not be simpler. To configure the settings, you will need your SIP ID that you received when registering with SIPNET as well as your password. Most softphones will prompt you for the following details only:
- SIP Server | Registrar | Server: sipnet.net
- Username | Login | User ID | Alias: YOUR SIP ID
- Password: YOUR PASSWORD
Other softphones may prompt you for other details:
- Outbound proxy (if required): sipnet.net
- Server | Proxy port: 5060 (UDP or TCP)
- Transport protocol: TCP или UDP (in the order of preference)
- Authorization name | Auth ID и т.п.: not used or specify SIP ID
- Display Name: any text, such as your actual name or account name; do not use Cyrillic script
- STUN server (if required): stun.sipnet.net, STUN port: 3478
Enjoy your communication!
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Call forwarding
You can forward your incoming calls automatically from SIP ID to another telephone number and always stay in touch.
It is a cost-effective solution if you move to another city or another country where you can access a land-line telephone (e.g. that of your family, friends, or colleagues) or use a local cellular operator's SIM-card to receive incoming calls.
You can specify a telephone number and call forwarding condition in the MyServices menu, Call Mangement page in Personal Office.