First let me say. After spending a significant amount of time debugging using gdb I can safely say this project is well engineered and the development team is top notch. Thank you for everything you have done. After much debugging and work, I am attempting to solve and document the issue of using multiple SIP lines for inbound and outbound calls with the same or different carriers. The information would need to show and explain an example using sofia.conf.xml, the XML dial plan bridging of a call and a corresponding JavaScript example. Perhaps I am the only one having this issue, but it would be nice to see examples with explanations of this within the wiki. Assuming I can finally solve my issue, I shall add the information myself. I realize that some of these facts are added already into the wiki, but the truth is that they are disconnected, contain minuscule amounts of information and make it hard for a FreeSwitch newbie such as myself to get my show up and running. Regards, Chris
As Anthony had mentioned in another thread, we have heard your ongoing complaints that there is not enough documentation, but we don't know what kind of documentation you really want. Please respond to this thread with what information that you were not able to find on the wiki, that would help you in using freeswitch. If no one responds, GREAT! Were done and don't need to document anymore. Otherwise, this is your opportunity to speak up and say what is missing.
Mike