Is it possible to set up a script to dial out from ASL3 to a SIP client? I’m looking for a CLI comand, rather than via RF / DTMF.
Where should I be looking - is this under the autopatch section?
I’m trying to set up an automated bridge where ASL3 will dial out to a SIP client (which will auto answer). And then write a script to check if there is an active call on SIP, and if not, dial out to re-establish the link.
Setup autopatch to call the SIP endpoint with PJSIP.
You can check for SIP calls easily enough with an asterisk command. Try “core show help pjsip” in the asterisk CLI for details. I don’t have a node up with PJSIP loaded at the moment to tell you exactly.
You can script making the call with an asterisk command. I would just do:
asterisk -rx "rpt fun [node] *61[SIP extension]"
or whatever the right command for your autopatch setup is. Then check for the call with another asterisk command. Put it all together in a shell script.
If you did this on a private node, the tx would not matter if it were a radioless node.
Connect ‘from the private node to the node you are listening in monitor mode’ and it will not trigger it’s tx…