Frequently Asked Questions

To obtain a digital number from VOIP.GE, you need to order the appropriate service and complete the required customer identification process.

To discuss available numbers and registration requirements, please visit our Contact page .

This depends on the type of number and its technical and regulatory status.

To determine whether a specific number can be used within a SIP infrastructure, the number and service details must be reviewed.

Yes. Companies can use multiple digital numbers within a unified business telephony infrastructure.

Numbers can be integrated with SIP Trunk, PBX or Call Center systems.

VOIP.GE services can be paid for through any bank in Georgia.

For online payments, you can also use our Online Payment page .

Yes. Service payments can be made by bank transfer.

Please contact us to receive payment details and information about the cost of a specific service.

Detailed information is available on our Online Payment page .

SIP Trunk is an IP-based telephony connection that allows a company's PBX system to connect to the telephone network over an IP network.

It is commonly used for business telephony, Call Centers and multi-channel voice communication.

For detailed information, please visit the SIP Trunk service page.

A dedicated Internet connection is not always required. However, network stability has a direct impact on VoIP quality.

SIP services can be used with an IP phone, a softphone application installed on a computer, or a PBX system.

Information about supported devices and configuration is available in our Documentation .

Both are IP-based business telephony systems designed to reduce the need for traditional on-site PBX infrastructure.

The specific architecture and service model depend on the requirements of the business.

Yes, provided that the existing PBX supports the required SIP functionality.

Integration can be implemented using SIP Trunk and the appropriate network configuration.

Yes. A PBX system can provide internal extensions, IVR, call forwarding, call groups and other business telephony features.

A dropped call can have several causes, including an unstable Internet connection, NAT, Firewall policies, SIP timeouts or network congestion.

Identifying the exact cause requires analysis of the network and the SIP session.

One-way audio is often related to RTP traffic, NAT or an incorrect Firewall configuration.

In such cases, SIP and RTP traffic should be checked in both directions.

A Codec determines how audio is encoded and decoded during a VoIP call.

Different Codecs have different bandwidth and processing requirements.

VOIP.GE services can be used with a wide range of SIP-compatible devices.

Supported manufacturers include Yealink, Grandstream, Fanvil and Cisco as well as other SIP-compatible manufacturers.

Configuration instructions are available in our Documentation .

Yes. With a compatible SIP softphone application, you can register and use a SIP account on a mobile device.

In this case, Internet connection quality and the device's network configuration are important factors.

To help us diagnose the problem quickly, please provide the following information:

For technical issues, you can contact us by phone or use our Technical Support page .

The more technical information you provide when describing the problem, the faster our team can diagnose it.

Where necessary, remote technical consultation and configuration checks may be provided.

The specific support method depends on the nature of the problem.