Hello.
I am trying to setup hello-world pjsip example as written in the wiki.
My pjsip.conf:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
auth=6001
aors=6001
[6001]
type=auth
auth_type=userpass
password=6001pass
username=6001
[6001]
type=aor
max_contacts=1
extensions.conf:
[from-internal]
exten => 100,1,Answer()
same => n,Wait(1)
same => n,Playback(hello-world)
same => n,Hangup()
Why my softphone is trying to connect to the asterisk, i see the following message in cli:
[May 22 10:22:17] NOTICE[951]: chan_sip.c:29060 handle_request_register: Registration from '"6001" <sip:6001@192.168.122.207>' failed for '192.168.122.1:5060' - Wrong password
So somewhy the call is processed by chan_sip.
If chan_sip is unloaded by module unload chan_sip
, then i see no messages in cli when softphone is trying to connect.
An there are even no open sip ports:
# netstat -ntlp
Active Internet connections (only servers)
Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name
tcp 0 0 127.0.0.1:5038 0.0.0.0:* LISTEN 879/asterisk
tcp 0 0 127.0.0.1:631 0.0.0.0:* LISTEN 861/cupsd
tcp 0 0 127.0.0.53:53 0.0.0.0:* LISTEN 499/systemd-resolve
tcp 0 0 127.0.0.54:53 0.0.0.0:* LISTEN 499/systemd-resolve
tcp6 0 0 ::1:631 :::* LISTEN 861/cupsd
Module res_pjsp is loaded, as shown by module show
:
res_pjsip.so Basic SIP resource 51 Running core
So the question is: what i forgot do do to enable test call via pjsip?
P.S.
$ asterisk -V
Asterisk 20.6.0~dfsg+~cs6.13.40431414-2build5