Hello,
I’m new with the usage of asterisk (Asterisk 18.2.0 currently running on ubuntu)
I’m trying to set a small call center for development and I’ve some trouble with the SIP OPTIONS
Asterisk send options and my softphone answer with 200 OK but when my softphone send option asterisk respond with a 401 Unauthoried
Here is pjsip.conf :
udp-transport]
type=transport
protocol=udp
bind=0.0.0.0
;===============ENDPOINT TEMPLATES
endpoint-basic ](!)
type=endpoint
transport=udp-transport
context=internal
disallow=all
allow=alaw
auth-userpass ](!)
type=auth
auth_type=userpass
aor-single-reg ](!)
type=aor
max_contacts=2
qualify_frequency=30
qualify_timeout=3.0
maximum_expiration=1800
;===============EXTENSION 6001
6001](endpoint-basic)
auth=auth6001
aors=6001
auth6001](auth-userpass)
password=6001
username=6001
6001](aor-single-reg)
there is a another thing that I don’t understant is why the nomber of sip options send by asterisk increase for exemple first login I receive 1 option every 30seconds I do a unregister/register and then I get 2 options from asterisk etc…
With wireshark there is a message : Suspected resend of frame: XXX
Thanks for help