I’ve got a working asterisk configuration for a Grandstream HT503. At least, it works at home. At work it fails with, "Got SIP response 503 “Service Unavailable” back from "
Looking at the SIP debugs, the thing that stands out is:
Home:
Sending to (no NAT)
Transmitting (no NAT) to
Work:
Sending to (NAT)
Reliably Transmitting (NAT) to
What is your local network at your home, what ip the 503 and the asterisk have?
What is your local network at your office, what ip the 503 and the asterisk have?
Also the network in your office is setup by you or a sysadmin?
The private IP is still the only advertised on your SDP.
Beside the localnet option, there are still a few parameters you need to tweak.
For example :
nat = force_rport,comedia
externaddr = 12.34.56.7 ;replace it with your public IP
externhost=foo.dyndns.net ; refreshed periodically
externrefresh=180 ; change the refresh
Check the section called NAT Support on the sip.conf sample configuration.
Your original problem’s description make reference to a 503 sip response, There is no such response on your sip trace. Beside that I see successfully established and media between
Got RTP packet from 192.168.1.120:5012 (type 00, seq 012433, ts 1663777460, len 000160)
Sent RTP packet to 192.168.1.47:7078 (type 00, seq 017204, ts 1663777456, len 000160)
Are you physically relocating the Asterisk System from Home to the Office and assigning a IP address to the asterisk system that is in the OFFICE Subnet 192.168.0.0/24? Or is it set to a static IP based on your Home Network Subnet 192.168.1.0/24?
As well if your IP address are correct for the network they are on make sure you are not binding asterisk to the wrong IP
ex: bindaddr=192.168.0.10
as well check your
localnet=
too make sure that it is on the correct network
Apologies for the slow reply - I completely missed it.
I am physically relocating my asterisk system - it’s just a VM atm.
In both environments, the VM gets a reserved address from a DHCP server.
I don’t think it can be binding to the wrong address as we can see communication between the asterisk server and the HT503 in the sip log.
As was pointed out earlier, the original problem was effectively solved, so I’m trying to put together a new question - please don’t let that stop you posting any further thoughts here though.