Hi, Several people have expressed an interest in how to configure Allstar to allow radio users to connect to the node-operator's local SIP phone. It turns out it's pretty easy (once you have a local SIP phone configured of course). To do this without configuring more general autopatch access with it's potential risks, you can create a specific context for just allowing a radio user to access one "outbound" SIP connection, the local SIP phone. In rpt.conf, I uncommented the "autopatchdn" function and duplicated and uncommented the "autopatchup" function. These two functions are in the [functions] stanza available to radio users. In the new "autopatchup" function, I changed the default DTMF string from 6 to 61 and added the "context=" option. I set the option to "context=node-op". So the function now looks like this: 61=autopatchup,context=node-op,noct=1,farenddisconnect=1,dialtime=20000 Then in extensions.conf, I added a stanza for [node-op]: [node-op] exten => 1,1,Answer exten => 1,n,Dial(SIP/200,10) exten => 1,n,Playback(vm-nobodyavail) exten => 1,n,Hangup Change the SIP/200 above to SIP/whatever-your-local-extension-is. Since we modified rpt.conf, you need to restart asterisk. Now, if the radio user keys in *611, the autopatch will be invoked and extension "1" in context [node-op] will be called where it will be answered and then will dial the local SIP phone extension. If you don't pickup, it will timeout, play the "nobody available to take your call" message, and hangup. Of course you can integrate this into a full autopatch configuration by modifying the default dialplan in context "radio" but I'm not willing to open my node up to dialing out to the world (yet). 73's, Keith KF7DRV