Hey, I tried using asterisk with GoTrunk with chan_sip but it is not supported anymore, so I tried using PJSIP but look at what its doing when I’m trying to login with my softphone
res_pjsip/pjsip_distributor.c:676 log_failed_request: Request ‘REGISTER’ from ‘sip:202@164.92.241.91’ failed for ‘81.65.133.166:33916’ (callid: 110164YjljMTNmMzAyNTg5MjJmMTZiMzkyNDFlODJkNmQwMTI) - No matching endpoint found
;=====CONFIGURE 2 EXTENSIONS FOR USE WITH SIP PHONE=====
[202]
type=endpoint
transport=transport-udp
context=from-internal
callerid= +44203xxxxxxx<+44203xxxxxxx> ; optional to set custom CID on Extension
disallow=all
allow=all
auth=202
aor=202
[202]
type=auth
auth_type=userpass
password=HIDDEN BUT I PUTTED SAME ON SOFTPHONE
username=HIDDEN BUT I PUTTED SAME ON SOFTPHONE
[202]
type=aor
mac_contacts=5
isn’t this good?
PS : I got this from gotrunk official website, it told me to put this in pjsip.conf
Please don’t tag me. I’ll respond if I have something to add and when I can. In this case I was outside on my bicycle, as it’s after my work hours.
Ensure that the endpoint actually exists by looking at “pjsip show endpoints”. If not then check to see if it failed to load for some reason by looking at logs when Asterisk starts.
It may be easier to additionally configure a FreePBX VM. You get valid PJSIP configurations with minimal input. Sometimes this may not be sufficient, but it is definitely a good starting point.
However, in general it is a bad idea to rely on cook book solutions without understanding them. Most of the chan_sip examples I see here are very badly done.