Hi I am new to asterisk, and learning as I plod along. Currently, I am trying to work out how to create a ring group without using AMP.
I set my inbound line to ring multiple lines by using Dial(SIP/101,SIP/102) but when I answered the call, the lines which didn't answer became locked with no dialtone, as though on a call. I thought that setting up a ring group might help, but could only find references to creating them through AMP. I downloaded and installed freePBX and all it's dependencies, shoe-horned it into my httpd.conf (my asterisk server is also my web server) and tried to log in. The setup page was empty apart from the header and footer, and checking the logs there were lots of references to "Undefined variable: amp_sections" and other undefined variables. Then I discovered that the freePBX installation had also overwritten my sip.conf and extensions.conf (which is shockingly bad, there should at least be a warning about that). All in all, not a good start :) any help would be MUCH appreciated thanks Chris _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
