I have an asterisk 16.6.2 installed behind NAT and my client is also behind NAT but when I am dialing 100 extension from peer 8000 as per dial plan call should run MOH application
but before that it is giving below error, any idea what am I missing?
[Dec 27 07:50:31] ERROR[30457]: res_pjsip.c:3460 ast_sip_set_tpselector_from_transport_name: Unable to retrieve PJSIP transport ‘UDP-NAT’
[8000]
type=endpoint
context=from-external
disallow=all
allow=ulaw;alaw;g729
transport=UDP-NAT
auth=8000
aors=8000
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
ice_support=yes ;This is specific to clients that support NAT traversal
;for media via ICE,STUN,TURN. See the wiki at:
;https://wiki.asterisk.org/wiki/x/D4FHAQ
;for a deeper explanation of this topic.
At startup does the console output show anything regarding the UDP-NAT transport? Does it appear in the CLI when you use the “pjsip show transports” CLI command?
Yes logs saying ->
[Jan 2 12:16:36] ERROR[25519] res_pjsip/config_transport.c: Transport ‘transport-udp-nat’ could not be started: Address already in use
[Jan 2 12:16:36] ERROR[25519] res_sorcery_config.c: Could not create an object of type ‘transport’ with id ‘transport-udp-nat’ from configuration file ‘pjsip.conf’
From the perspective of PJSIP debug won’t really change it. We ask the system to bind to “0.0.0.0 port 5060” and it merely says something is already listening, it doesn’t state what. You have to figure that out yourself. For example: Is chan_sip loaded? That will by default bind to it.