Hi list, BTW, great to see your group at LinuxFest. It was my first time out there, but it won't be my last.
I've managed to hack a few Panasonic JOIP/Globarange phones so that they are freed from there original service provider and now work for me on a standard SIP network with Asterisk. I did this before on another computer and had it working well with outbound lines and everything. Now I can't even get the extensions to talk to each other. When I try I only get "That number is not valid" or "The number you have dialed is not in service." All the extensions appear on the flash panel, and I was able to have time and echo test working, though I think I recently broke those too. Do I need to set up a trunk or incoming or outgoing routes in order to get the various extensions to talk to each other? If there are any tricks that I am missing I'd love to hear them. I don't want to post all the config files and ask you guys to debug it for me. I just want to know if there are some basic principles i need to apply to get a simple set of extensions working. I've so far not found any faqs to this end. Thanks darryl --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
