im using Asterisk 16.17.0 on ubuntu 20.04.2 LTS
ok i install libsrtp , and then i reinstall Asterisk (./configure; make; make install) .
I have also ensured that the SRTP module is selected in the Asterisk in the menuselect
I configure tls-sip on my server asterisk, my SIP users that is successfully registered .
the actual PJSIP configuration
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
cipher=ADH-AES256-SHA,ADH-AES128-SHA
method=tlsv1
[6006]
type=aor
max_contacts=1
;remove_existing=yes
[6006]
type=auth
auth_type=userpass
password=0001
username=6006
[6006]
type=endpoint
aors=6006
auth=6006
context= work
disallow=all
allow=ulaw
allow=alaw
allow=gsm
transport=transport-tls
media_encryption=sdes
use_avpf=yes
[6007]
type=aor
max_contacts=1
; remove_existing=yes
[6007]
type=auth
auth_type=userpass
password=0000
username=6007
[6007] type=endpoint
aors=6007
auth=6007
context= work
disallow=all
allow=ulaw
allow=alaw
allow=gsm
transport=transport-tls
media_encryption=sdes
use_avpf=yes
the SIP traces using “pjsip set logger on”
sip trace.txt (38.9 KB)
When trying to make a call, I am getting the below error:
ERROR[2650]: res_pjsip_session.c:936 handle_incoming_sdp: 6007: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
and this is my softphones settings
