Need help in troubleshooting SIP Trunk with IP Based Auth

Hi, Need help pls, My SIP provider gave the the ff details to be configured in my Asterisk Server

LAN IP: 172.16.1.98/23
SBC IP: 10.211.3.12
SIP DN Block 0885250001

My Asterisk Server IP is 172.16.1.99

in sip.conf i have

[trunk-in]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=10.211.3.12
dtmfmode=rfc2833
context=inbound
canreinvite=no
allow=alaw
allow=ulaw
allow=g729
allow=gsm

in my extensions.conf i have

exten => 0885250001,1,Answer()
exten => 0885250001,n,Dial(SIP/103,30,r)
exten => 0885250001,n,Hangup()

Now, When I try to do an inbound call by dialing 0885250001 I can see in the console that SIP/103 is ringing but on the phone i used to call the number above I can hear anything, after a few sec the call just drops

I have attached the debug file i got. Im guessing this is a codec problem, and I have already installed g729.so

debug.txt (49.7 KB)

Any help is appreciated. Thanks

You can’t hear any voices. maybe that’s a NAT or firewall issue.
check your sip.conf about NAT or check firewall!!

Hi @hsunryou any specific issue on NAT and Firewall?

Although not relevant to the problem, insecure=invite does nothing if you only have IP authentication, and canreinvite has been called directmedia for most of the last decade.