NAT, RTP, PJSIP. No audio problem in Asterisk 20

Asterisk Asterisk SIP
1

Hi! I’ve got the same problem. Could someone help me please.

This is my pjsip.conf:
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0:5020
allow_reload=yes
;NAT SETTINGS:
local_net=192.168.0.0/24
external_media_address=public_ip
external_signaling_address=public_ip

codecs
disallow = all
allow = gsm
language = es

softphones
type = endpoint
context = muundogando
allow = gsm
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733
direct_media = no
transport = transport-udp-nat
;media_address = public_ip
;media_use_received_transport = yes
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes

[auth-userpass]
type = auth
auth_type = userpass

[aor-single-reg]
type = aor
max_contacts = 1
remove_existing = yes

[100]
auth = 100
aors = 100
callerid = 100 <100>

[100]
password = 100
username = 100

[100]
mailboxes = 100@example

[1]
auth = 1
aors = 1
callerid = 1

[1]
password = 1
username = 1

[1]
mailboxes = 1@example

And this is my extensions.conf:
[general]

[muundogando]
exten => 1,1, Answer()
same => n, NoOp(Valor del argumento: ${id_vaca})
same => n, Playback(vaca-fuera-perimetro-1)
same => n, SayDigits(${id_vaca})
same => n, Playback(vaca-fuera-perimetro-2)
same => n, Playback(demo-echotest)
same => n, Hangup()

Im using Linphone as a SIP contact. In my asterisk everything seems to work but… there is no audio

2

This is a firewall issue NOT asterisk.
What firewall do you have in place?
Also make sure you have the appropriate ports opened in your firewall system.
10000-25000 rtp
5060 udp
5061 udp
& any other port you are forwarding to your sip service.

3

You are missing “type=…” in several sections, and I don’t think there is an option called “password”

4

The configuration is like this, but the web do not allow me to put it on the right way.
image

5

Okey, I have changed the rtp.conf to:
image

I’ll open the ports and try again.
Thanks!

6

Use the </> button.

7

I open de ports in the router and it function!!
Thanks!!!

1 Like
8

try: 10000 - 25000
with your current settings of 10000 - 10002
Your not going to get full access to corresponding ports

9

This will only allow one call leg, so no through calls, and I’d expect calls to fail if they followed on too close after a completed call.
The 10002 should never get used, as it would need 10003, for RTCP and that is not available.