Need help in troubleshooting SIP Trunk with IP Based Auth

Asterisk Asterisk SIP
1

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

2

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

3

Hi @hsunryou any specific issue on NAT and Firewall?

4

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.

5

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