SIP TLS TRUNK between 2 asterisk server

Do you mean i could still keep the certificate validation even without registration?
Because that is all what i am looking for.

hello guys, i made some modifications and this is what i am getting right now:

AST1 of ip address 192.168.133.100
AST2 of ip address 192.168.133.99

AST1 pjsip.conf:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0


[transport-tls]
type=transport
protocol=tls
bind=192.168.133.100:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
ca_list_file=/etc/asterisk/keys/ca.crt
ca_list_path=/etc/asterisk/keys
method=tlsv1_2
require_client_cert=no
verify_client=no
verify_server=no

AST2 pjsip.conf:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0


[transport-tls]
type=transport
protocol=tls
bind=192.168.133.99:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
ca_list_file=/etc/asterisk/keys/ca.crt
ca_list_path=/etc/asterisk/keys
method=tlsv1_2
require_client_cert=no
verify_client=no
verify_server=no

[trunk]
type=aor
contact=sip:192.168.133.100:5061

[trunk]
type=endpoint
context=phones
allow=gsm,alaw,ulaw
aors=trunk
direct_media=no
transport=transport-tls

[trunk]
type=identify
endpoint=trunk
match=192.168.133.99

AST2 extensions.conf:

exten => _02X.,1,Dial(PJSIP/${EXTEN:2:}@trunk)
same =>      n,Hangup()

Now i am trying to make a call from a sip account on AST2 to a sip account on AST1

and this is what i am getting at AST2:


[Mar  1 07:41:41] WARNING[1645]: pjproject: <?>:                   SSL SSL_ERROR                                                                                       _SSL (Read): Level: 0 err: <67567754> <rsa routines-RSA_padding_check_PKCS1_type                                                                                       _1-invalid padding> len: 65535 peer: 192.168.133.99:34417
[Mar  1 07:41:41] WARNING[1645]: pjproject: <?>:                   SSL SSL_ERROR                                                                                       _SSL (Read): Level: 1 err: <67530866> <rsa routines-rsa_ossl_public_decrypt-padd                                                                                       ing check failed> len: 65535 peer: 192.168.133.99:34417
[Mar  1 07:41:41] WARNING[1645]: pjproject: <?>:                   SSL SSL_ERROR                                                                                       _SSL (Read): Level: 2 err: <218910726> <asn1 encoding routines-ASN1_item_verify-                                                                                       EVP lib> len: 65535 peer: 192.168.133.99:34417

NOTE:
to be clear at each asterisk server i made their own certificates using the script!

NOTE:

if i make these modifications to trunk then the call goes fine!!

[trunk]
type=aor
contact=sip:192.168.133.100:5060

[trunk]
type=endpoint
context=phones
allow=gsm,alaw,ulaw
aors=trunk
direct_media=no
transport=transport-udp

[trunk]
type=identify
endpoint=trunk
match=192.168.133.99

Seems pretty pointless having TLS, then.

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i am just trying it to figure out what is the issue!!!

You never authenticate the caller. Even for the callee, the certificate could be signed by the North Korean Ministry of Intelligence, for all you care.

1 Like

I already said that i am playing with the configurations just to try figuring out what is the issue.

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