Greetings all,
A few months back, I started upgrading my nodes from the older ASL software to ASL-3. I'm catching up on some programming work I've been meaning to do, but have hit a wall on setting up SIP devices in the newer PJSIP framework. For background, I have a hub node that runs in the Amazon AWS cloud (49935), which links multiple repeaters and simplex nodes across SoCal, Southern Nevada, and Arizona. I've typically used SIP for remote access via Zoiper, but also so we can use SIP phones, generally Polycom devices, as desktop stations. I also have SIP phones at two of our repeater sites due to their remote location.
In my previous configuration, both the hub node (49935) and my main server at home hosted the SIP connectivity and were trunked together via IAX.
The wall I'm hitting right now is with the hub node in the cloud. I cannot for the life of me get audio across the path from the server to the endpoint. From what I've seen so far, I believe this is a problem due to NAT somewhere down the line. However, while I had this accounted for in the old SIP configuration, I cannot for the life of me find instructions on how to deal with it in PJSIP that I can understand and that seem to work.
Software version info is below.
Thank you in advance to whoever may be able to help me out on this one!
Tom, KI6GOA
OS : Debian GNU/Linux 13 (trixie)
OS Kernel : 6.12.74+deb13+1-cloud-amd64
Asterisk : 22.7.0+asl3-3.7.1-1.deb13
ASL [app_rpt] : 3.7.1
Installed ASL packages :
Package Version
============================== ==============================
allmon3 1.7.0-1.deb13
asl-apt-repos 2.0-1.deb13
asl3 3.15-1.deb13
asl3-asterisk 2:22.7.0+asl3-3.7.1-1.deb13
asl3-asterisk-config 2:22.7.0+asl3-3.7.1-1.deb13
asl3-asterisk-modules 2:22.7.0+asl3-3.7.1-1.deb13
asl3-menu 1.16-1.deb13
asl3-update-nodelist 2.0.0-1.deb13
dahdi 1:3.1.0-2.1
dahdi-dkms 1:3.4.0-10.asl.deb13
dahdi-linux 1:3.4.0-10.asl.deb13
dahdi-source 1:3.4.0-10.asl.deb13