pstn to sip gateway

============================================== View hardware configuration # Dahdi_hardware==============================================View Dahdi Service ConfigurationMore/etc/dahdi/system.confShow the following content, obviously less my PSTN card configuration# Global DataLoadzone = usDefaultzone = usRebuilding the Dahdi service configuration#dahdi_genconfView Dahdi Service Configuration again# more/etc/dahdi/system.confShow# autogenerated by/usr/

At the request of the dynaguy brothers, I have attached a relatively complete experiment to colleagues who are still exploring: (There are some problems with the 2.0beta experiment, so I will demonstrate it with the most stable 1.2.3) In this test, we did not discuss the issue of using the SIP Trunk directly to connect other sip servers without board installation to enable the

Introduction 1. About the SIP protocol Currently, there are three basic communication protocols based on VOIP: H 323, SIP, and MGCP. The following describes the comparison between the H 323 protocol and the SIP protocol. H.323 was proposed by ITU. It attempts to regard IP phones as well-known traditional phones, but the transmission mode is changed from circuit s

If you have a SIP account from a carrier, you can configure the SIP to dial an external phone. The SIP account (or the device providing the account) is referred to as the SIP gateway in FreeSWITCH. Adding a gateway only needs to c

This article from csdn ucser, http://blog.csdn.net/perfectpdl reprinted to indicate the source, thank you! I have created a freeswitch learning and communication group, 45211986. welcome to join. Freeswtich can be used as the rtmp and SIP gateway of the Streaming Media Protocol. It can communicate with the SIP video phone through flash in a web browser. Th