Thanks everyone for your help! I think I have it working at this point. Adding --cc "rtp-peer 192.168.3.225" did the trick and I could call into my pbx and listen to a recording. I still need to make sure incoming calls work and ring the amps phone (which I'll attempt this week) but I have no reason to believe it wouldn't work as expected.

Andreas noted correctly that there is no NAT in between my machine and the asterisk server. If we were to suppose that there was a NAT and the asterisk server lived somewhere on the Internet, would I need to take any other considerations aside from using my external IP address in the osmo-cc-sip-endpoint arguments in place of the internal IP? I'm just thinking about a future scenario if I want to take the phone somewhere else for testing.

Thanks,
Mike.

On Sun, Feb 26, 2023 at 2:54 AM Andreas Eversberg <andreas@eversberg.eu> wrote:
Hi Mike,

you need to specify your RTP peer, if the remote RTP endpoint is on a
different machine. The default is 127.0.0.1, which the remote SIP
endpoint @192.168.5.228 reach. As 192.168.3.225 is your local network
interface to talk to the other SIP endpoint, you should add --cc
"rtp-peer 192.168.3.225":

amps --limesdr-mini --channel 334 - channel 332 -o --fast-math --cc
"rtp-peer 192.168.3.225"

Note that if you define STUN server or public IP at
osmo-cc-sip-endpoint, the "rtp-peer" gets automatically translated into
public IP address. But in your case there seems to be no NAT in between.

I use 'sngrep' to check that by looking at the INVITE message.

Regards,

Andreas


On 2/23/23 04:34, Famicoman wrote:
> I am running amps with:
> $ amps --limesdr-mini --channel 334 - channel 332 -o --fast-math
>
> I am running osmo-cc-sip-endpoint with:
> $ osmo-cc-sip-endpoint --local 192.168.3.225 --remote 192.168.5.228 -R
> 8927188@192.168.5.228 -A 8927188 secretpasswordhere 192.168.5.228 -v 0 *
--
osmocom-analog mailing list -- osmocom-analog@lists.osmocom.org
To unsubscribe send an email to osmocom-analog-leave@lists.osmocom.org