Why is the call not going through?

Question

Asterisk

Good afternoon. I am just starting to study this system, I used to work with freepbx, elastix, now I decided to try pure asterisk (version 13).
Added two sip numbers to pjsip.conf:

spoiler

[transport-udp]
type=transport
protocol=udp    ;udp,tcp,tls,ws,wss
bind=0.0.0.0

[101]
type=endpoint
context=from-internal
disallow=all
allow=alaw
transport=transport-udp
auth=auth101
aors=101

[auth101]
type=auth
auth_type=userpass
password=101
username=101

[101]
type=aor
max_contacts=1

[102]
type=endpoint
context=from-internal
disallow=all
allow=alaw
transport=transport-udp
auth=auth102
aors=102

[auth102]
type=auth
auth_type=userpass
password=102
username=102

[102]
type=aor
max_contacts=1

Added to extensions.conf:

[from-internal]
exten => 101,1,Dial(SIP/101)
exten => 102,1,Dial(SIP/102)

Phones registered, but when calling from one number to the second - short beeps.
What am I doing wrong?

Answer 1

Insist PJSIP and call SIP
https://wiki.asterisk.org/wiki/display/AST/Dialing...