First I just wanted to say hi, this is my first post. I look forward to learning a lot abou VoIP and Asterisk over the next year.
Secondly, I’m having some issues using asterisk with a modified SR Gizmo. I have setup my sip, extensions, and iax.conf files and have the asterisk server running.
I have a dial tone, but no monkeys!
My gizmo WAN has been modified to have an IP on the range of my os x box, as its server to get info from, with autoconfiguration turned off.
Sip has been configured with the OS X box’s IP/Domain, and Line1 created with a user/phone num, caller id name, auth. name, and a password. This line is showing up as configured after having it connect to the asterisk server.
“sip show settings” results can be provided upon request, as well as the debug info spit out when a phone call is attempted.
Please let me know if anyone has any advice or experience with this. Thanks
Edit: I would also like to note, that I do not intend on using SR service, but instead my own internal. self contained VoIP network. Thanks and sorry for any confusion.