[App_rpt-users] Calling into the node operator's local SIP phone

Keith Williamson hkwilliamson at gmail.com
Sun Feb 7 14:29:03 EST 2010


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:


Then in extensions.conf, I added a stanza for [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

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).


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