A learning plan to build telephone system with openser

(Practice my poor English....)

Yesterday, I find a good document for openser introduce book in eMule search machine.

The book filename is Packt.Building.Telephony.Systems.with.OpenSER.Apr.2008.pdf

Some discussions in Internet has mentioned the server will share some bandwidth when talking in asterisk system. It seems the asterisk doesn't redirect peers to let them become peer-to-peer. There is such function in openser. It can be referred in official website. However, the openser seems doesn't support the voice service such as Voice mail, IVR and etc.

Therefore, I think to build up main SIP server by OpenSER will be a good way and use asterisk to be the openser gateway to provide voice service and etc.

Although I have compiled the OpenSIPs rather than OpenSER, I think the methods should be the same. It will be a good chance to learn the telephone system with this book. :D