Hi,
Got problems with incoming SIP calls.
Scenario:
Server1: 3cx or any other server
Server2: Asterisk 16.2.1 . PJPROJECT 2.8
Server2 registers on Server1 with SIP ID 1121.
Registration is OK.
Server2 outgoing calls are OK.
INVITE, unauthorized, INVITE with password, OK, RINGING,...
Troubles with incoming calls / incoming INVITE's .
I can not identify endpoint by IP, I have multiple registrations on the
same Server1.
As far as I understood, res_pjsip_endpoint_identifier_user match
endpoint by "From" header, so it will not match also.
match_headers also seems useless (not able to match "INVITE" string,
just headers like "TO:").
Is there any way to match incoming INVITE calls ? (also OPTIONS, NOTIFY,
... packets)
It should be a typical scenario, but it does not work...
Is there any way to make it working ?
[0.0.0.0-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[endpoint0](!)
type=endpoint
transport=0.0.0.0-udp
disallow=all
allow=alaw
allow=ulaw
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
t38_udptl_nat=no
dtmf_mode=auto
direct_media=yes
from_domain=172.16.25.23
timers_sess_expires=1800
tone_zone=ru
language=ru
rewrite_contact=yes
rtp_symmetric=yes
force_rport=yes
[registration0](!)
type=registration
transport=0.0.0.0-udp
retry_interval=60
max_retries=10
expiration=3600
auth_rejection_permanent=yes
server_uri=sip:172.16.25.23
[fxs17](endpoint0)
context=from-sip-fxs
aors=fxs17
outbound_auth=fxs17
from_user=1121
set_var=DAHDICHAN=17
[fxs17]
type=aor
qualify_frequency=60
contact=sip:1121@172.16.25.23
[fxs17]
type=auth
auth_type=userpass
password=111111
username=1121
[fxs17](registration0)
outbound_auth=fxs17
client_uri=sip:1121@172.16.25.23
contact_user=fxs17
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