I'm using a Netvanta 6355 and working on setting it up as a transparent proxy but when I have the configuration in place that I copy from here: https://supportforums.adtran.com/docs/DOC-7023?q=Issues%20with%20Transparent%20SIP%20Proxy the phones will not register. Below is are pertinent parts of the config.. I will see the phones show up when I do a "show sip proxy reg" but show as unregistered. The primary issue right now is that when I have this config in place, the phones cannot register with our PBX through our SBC, I have to disable sip proxy before registrations will work properly.
domain-lookup database local
name-server 184.108.40.206 220.127.116.11 18.104.22.168
ip dhcp pool "Voice"
network 10.10.83.0 255.255.255.0
dns-server 10.10.83.1 22.214.171.124
interface vlan 1
no ip address
interface vlan 100
ip address 10.10.83.1 255.255.255.0
ip access-policy Private
media-gateway ip primary
sip udp 5090
sip udp 5360
no sip tcp
voice feature-mode network
voice forward-mode network
voice dial-plan 0 local NXX-XXXX
voice dial-plan 1 extensions 05XX
voice trunk T01 type analog supervision loop-start
description "FXO Failover"
connect fxo 0/1
rtp delay-mode adaptive
voice grouped-trunk FXO
accept 911 cost 0
accept 9NXX-XXXX cost 0
accept 9NXX-NXX-XXXX cost 0
accept NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
voice ring-group 0599
voip name-service host mast.3cx.us sip udp
voip name-service host voice.network sip udp
sip proxy transparent
sip proxy failover accept-registrations
sip proxy failover match-digits 4
I ended up opening a case with support and it was found to be an issue with how the Yealink phones (also one model of Grandstream I tested) responded to the SIP 407 Proxy Authentication Required message they were receiving from the Adtran. The phones never resubmitted their register request with the credentials, just continued to send in register requests without them.
Because of this I ended up going a different direction than the SIP transparent proxy feature (phones are registered to the PBX AND to the Adtran on as a secondary SIP server for survivability). So this can be marked as completed/answered
Do you mind sharing some info on this? I am trying to do the same thing with Digium SWVX and a 908e with users connected over a vpn?