Mike, Not sure what the exact problem is, you did not state it. I'm guessing your problem is in trunking/peering the asterisk/home-pbx to asterisk/app_rpt You are missing a context= statement. Probably want to use radio-control for it on app_rpt side. If not, asterisk knows not what stanza to associate the call with and you will not get the correct channel. There is more than one way to do the trunking or peering systems. I'm not sure I am following what you are doing from what is shown. ...mike/kb8jnm
-------- Original Message -------- Subject: [App_rpt-users] reverse autopatch for IP phone network From: Mike Lussier <mike.lussier@gmail.com> Date: Sat, December 27, 2014 10:39 am To: ALL STAR <app_rpt-users@ohnosec.org>
Good morning all.
I did my due diligence by researching the archives for my answer first.
I saw that K1LNX had posted in 2010 [App_rpt-users] Example of how to setup a sip phone for reverse autopatch. I have been playing with this the past 24 hours. My home pbx is asterisk PBX and my nodes are all Beagle Bone Black running the new load. It seems to be a problem with sip. I have configured sip trunk (radio1) on my PBX peer details for: Host=10.133.54.132 Username=ae4ml Secret=passwd type=friend
registration string ae4ml:passwd@10.133.54.132/radio1
Outbound route radio1 dialplan 1XXX Trunk radio1
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