No sound on sip call

Asterisk Asterisk SIP
1

I installed asterisk 16 on aws ubuntu server.

I only edited two files which is sip.conf and extensions.conf

see my editing…

[sip.conf]

[1003]
type = peer
secret = 1003
port = 5060
autocreatepeer = no
context = outbound
disallow = all
allow = ulaw
dtmfmode = auto
host = dynamic
nat = force_rport,comedia
canreinvite = no
quality = yes
insecure = port, invite
avpf = yes
localnet = 172.31.46.149
externip = 13.125.195.45
[1004]
type = peer
secret = 1004
port = 5060
autocreatepeer = no
context = outbound
disallow = all
allow = ulaw
dtmfmode = auto
host = dynamic
nat = force_rport,comedia
canreinvite = no
quality = yes
insecure = port, invite
avpf = yes
localnet = 172.31.46.149
externip = 13.125.195.45

[extensions.conf]

[outbound]
exten => _1XXX,1,Ringing
exten => _1XXX,n,Dial(SIP/${EXTEN})
exten => _1XXX,n,Hangup

However, when I make a call both side using linephone?
call is established but I can’t hear any sound.
I opened 5060 port.

when I check my sip info lookes like below

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1001/1001 (Unspecified) D Yes No 0 Unmonitored
1002/1002 (Unspecified) D Yes No 0 Unmonitored
1003/1003 119.203.28.9 D Yes Yes 44308 Unmonitored
1004/1004 119.203.28.9 D Yes Yes 39502 Unmonitored

What do I have to do to hear sound during calls?

2

localnet and externip are global, so must be in general.

localnet is a network, so should include a netmask.

Without knowing your network configuration, we cannot validate even a correctly formed localnet.

3

Thanks for your advice.
I put my localnet with net mask and externip at general
and opened all udp port.
finally I could hear the sound…

4

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