Inbound call failed in sip.conf

[htrakotoarimanana]

Hello,
I have a trunk sip configured in the file sip.conf. We use ovh ligne

but when I call the number,the call is ended after around 30s,and call ended
In softphone I catch those errors:

not registered (code: 403): the must frequent
Registration failed due to STUN server (DNS resolving or connection) error (code: 59)

And in the side of the server which contains asterisk I can catch this error
Retransmission timeout reached on transmission 21539149707002376716095 for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
and sometimes there is nothing and the call is ended after 1s

We hase Asterisk certified/16.8-cert2. May be there is parameters missing or wrong configuration

Below the configuration in the sip.conf

[general]
udpbindaddr=0.0.0.0
directmedia=no

; OVH
defaultexpiry=1800
bindport=5060
bindaddr=0.0.0.0
srvlookup=no ;
register => [number]:[password]@sip.ovh.fr:5060

[ovh]
type=peer
username=[number]
secret=[password]
host=sip.ovh.fr
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw,alaw,gsm,g722,g729
insecure=port,invite
fromdomain=sip.ovh.fr
context=trunk-sip
language=fr
rtp_symmetric=yes
nat=yes

Thank a so much

[david551]

nat=yes is not a defined value. More generally, go back to first principles, as you have deprecated and overused options.

You have no options to allow Asterisk to work behind NAT. nat= is not sufficient, and not always necessary. I don’t understand why it is complaining abut stun servers when none is configured.

[htrakotoarimanana]

Thank you
could you give me what paramaters did you mean as depredated please

[david551]

canreinvite.

nat=yes

rtp_symmetric (not sure if this one ever existed for chan_sip.