PJSIP contact unavailable with private ip

Asterisk Asterisk SIP
1

Register packet of my endpoint

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image1166×281 20.8 KB

Invite packet to my enpoint

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image1433×512 36.1 KB

My configuration

[global]
max_forwards=70
default_realm=acedial
keep_alive_interval=300
directmedia=no
rewrite_contact=yes
rtp_symmetric=yes

[udp_transport]
type=transport
protocol=udp
bind=0.0.0.0:5062

[wss_transport]
type=transport
protocol=wss
bind=0.0.0.0

[tcp_transport]
type=transport
protocol=tcp
bind=0.0.0.0:5062

[tls_transport]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/ssl/cert.crt
priv_key_file=/ssl/private.key
method=sslv23

[single_aor](!)
max_contacts=1
;qualify_frequency=120
remove_existing=yes

[userpass_auth](!)
auth_type=userpass

[basic_endpoint](!)
context=from-extensions

[phone_endpoint](!)
allow=ulaw,alaw,g722,gsm,vp9,vp8,h264

[92331](basic_endpoint,phone_endpoint)
type=endpoint
Callerid=“Hamza” <92331>
auth=92331
aors=92331

[92331](single_aor)
type=aor

[92331](userpass_auth)
type=auth
username=92331
password=pass

[92332](basic_endpoint,phone_endpoint)
type=endpoint
Callerid=“Ali” <92332>
auth=92332
aors=92332

[92332](single_aor)
type=aor

[92332](userpass_auth)
type=auth
username=92332
password=pass

What is wrong with the configuration?
Even user is registered but error enapoint registered no route to destination.
any help thanks.

2

I can’t see anywhere in the documentation that mentions any general section for pjsip.conf. In fact https://www.asterisk.org/pjsip-configuration-design/ says there isn’t one. As such, I would say that your rewrite_contact line is being ignored.

3

Thanks for the reply. you are saying that I need to put all global context values to from-extensions context?

4

Thank you save my time. I moved rewrite_contact and direct_media=no to phone_endpoint and then call working fine

5

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