Sin audio en llamadas

Hola a todos; espero que esten bien.

Busco la ayuda de ustedes por el siguiente tema:

En google cloud instale un debian 11 , dentro de este tengo instalado asterisk Asterisk 20.10.0.

Google Cloud me entrega:

Internal IP 10.128.0.26
External IP 34.42.181.37


En asterisk configure mi archivo PJSIP.CONF:

;================================ TRANSPORTS ==
[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0
external_media_address = 34.42.181.37
external_signaling_address = 34.42.181.37
local_net = 10.128.0.0/20

;================================ CONFIG FOR SIP ITSP ==
[dcs-trunk]
type = registration
outbound_auth = dcs-trunk-auth
server_uri = sip:sip.digiumcloud.net
retry_interval = 60

[dcs-trunk-auth]
type = auth
auth_type = userpass
username = myaccountID
password = ASTRONGPASSWORD

[dcs-endpoint]
type = endpoint
context = DCS-Incoming
allow = !all,g722,ulaw
outbound_auth = dcs-auth
aors = dcs-aor
direct_media = no
from_domain = sip.digiumcloud.net

[dcs-auth]
type = auth
auth_type = userpass
username = myaccountID
password = ASTRONGPASSWORD
realm = sip.digiumcloud.net

[dcs-aor]
type = aor
contact = sip:sip.digiumcloud.net

[dcs-identify]
type = identify
endpoint = dcs-endpoint
match = hotkey404-asterisk-gcp.ip-dynamic.org

;================================ ENDPOINT TEMPLATES ==
endpoint-internal-d70
type = endpoint
context = Long-Distance
allow = !all,g722,ulaw
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733

auth-userpass
type = auth
auth_type = userpass

aor-single-reg
type = aor
max_contacts = 20

;================================ Usuario Tw0 (1100) ==
1100
type = endpoint
auth = 1100-auth
aors = 1100
callerid = Tw0 <1100>
allow = ulaw,g722

1100-auth
type = auth
username = 1100
password = 1100

1100
type = aor
mailboxes = 1100@example
qualify_frequency = 30

;================================ Usuario Tw1 (1101) ==
1101
type = endpoint
auth = 1101-auth
aors = 1101
callerid = Tw1 <1101>
allow = ulaw,g722

1101-auth
type = auth
username = 1101
password = 1101

1101
type = aor
mailboxes = 1101@example
qualify_frequency = 30


tengo creado mi regla de firewall:

34.42.181.37|tcp:5060

udp:5060, 10000-20000|
| — | — | — | — |


archivo rtp.conf:

[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;


las extensiones SIP se logran conectar en telefonos , pero no tengo audio de mis llamadas.

podrian orientarme que tema podria estar faltano, e realizado varias pruebas pero sigue igual

Spanish is my native language, but in order to maintain consistency with the forum’s language, I will post my reply in English. I think it won’t be hard for you to translate it using an AI tool and reply in English with the same AI tool.

Now, to address the main issue, please add these options to your templates and trunks:

  • rewrite_contact=yes
  • rtp_symmetric=yes
  • force_rport=yes

Also, make sure to inspect the RTP traffic using an RTP debug.