Yate Is Free VOIP Client For Linux That Supports Gtalk.

Saturday, August 20, 2011

Having worked as a system administrator in a call center environment, I have learned that finding a free VOIP (Voice Over IP) software that actually works is quite difficult. The alternative to that can cost anything from 20$-35$ per month for each username rented from the service provider.

This also includes the server, client and supported usernames for managing and using a VOIP setup. Unfortunately, there are very few VOIP clients for Linux. And an even fewer are free.

Yate (Yet Another Telephony Engine) is a VOIP telephony with flexible routing engine which can reduce infrastructure costs for call center businesses. However, Yate is not merely designed for call centers and can be used as a VOIP client or server by common users who use voice over IP and businesses who require making a lot of calls.
In the protocols section you can enter a sip or use jabber, h323 or iax. Yate can be used as a VoIP server, VoIP client, Conference server for up to 200 channels in a single conference, VoIP to PSTN gateway, PC2Phone and Phone2PC gateway, IP Telephony server and/or client with:

H.323 gatekeeper
H.323 multiple endpoint server
H.323<->SIP Proxy
SIP session border controller
SIP router
SIP registration server
IAX server and/or client
Jingle client or server
MGCP server (Call Agent)
ISDN passive and active recorder
ISDN, RBS, analog passive recorder
SS7 switch
Call center server
IVR engine
Prepaid and/or postpaid cards system

A number of accounts can be created and added from the Accounts section. Yate supports FWD, iptel,sipgate, Gtalk, ekiga and voxgratia. Just enter your account credentials to add a username to yate.

After selecting a protocol from the Calls menu, enter a number and click Call to dial.

Calls can also be made from the address book (Contacts menu) by selecting added contacts. Contacts can be added, deleted and called by selecting the respective buttons from the Contacts menu.

A detailed call log is automatically preserved in the Call Log menu and can be used for redialing numbers.

You can enable a ringer, enable auto answer, force active the last outgoing call, etc from the Settings. Domain credentials can also be added from the same tab.