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I have an Asterisk server sitting on my network behind a pfSense firewall, it has two trunks, one for my household provided by my ISP using PJSIP and the other for my business provided by a third party which use plain SIP.

Dialing out works with no issues, however when getting calls on the household trunk (PJSIP) the caller can not hear me until they speak or make some noise.

Here is my PJSIP configuration:

[transport-udp]
type=transport
allow_reload=yes
protocol=udp
bind=0.0.0.0:5060
local_net=192.168.2.0/24
external_media_address=REDACTED
external_signaling_address=REDACTED

[net]
type=registration
transport=transport-udp
outbound_auth=net
server_uri=sip:REDACTED
client_uri=sip:REDACTED@REDACTED
retry_interval=30
forbidden_retry_interval=30
fatal_retry_interval=30
auth_rejection_permanent=no
max_retries=10000
expiration=30
line=yes
endpoint=net
contact_user=REDACTED

[net]
type=auth
auth_type=userpass
username=REDACTED
password=REDACTED

[net]
type=aor
contact=sip:REDACTED:5060
maximum_expiration=30

[net]
type=endpoint
language=au
transport=transport-udp
context=from-net
disallow=all
allow=g722
allow=ulaw
allow=alaw
allow=g729
outbound_auth=net
aors=net
moh_suggest=music
trust_id_inbound=yes
from_user=REDACTED
from_domain=REDACTED
direct_media=no
rewrite_contact=yes

[net]
type=identify
endpoint=net
match=REDACTED

Since there are two SIP providers the call is matched based on the IP addresses provided by my ISP of their SIP servers.

How do I fix this?

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  • Dialing out works with no issues, however when getting calls on the household trunk (PJSIP) the caller can not hear me until they speak or make some noise. This issue is quite rare, did you check the RTP ports config in the pfSense? Did you use the SIP provider with other method to check it's not a provider issue? Apr 19, 2020 at 22:37

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