I would like to know how I should setup a SIP trunk without registration just for sends qualify options. I use it on Asterisk 13 with the same settings and works well just setting up data on tables ps_aors, ps_endpoints and ps_endpoint_id_ips. Settings: mercury-telecom-01*CLI> pjsip show aor GTGRO…

A reload should trigger it to happen if I recall correctly, or an inbound REGISTER. This is because otherwise you’d need to be constantly checking the database and reconciling the state.

On the aor section the qualify_frequency Interval between attempts to qualify the contact for reachability. If 0 never qualify. Time in seconds. So I think with endpoint and aor section is enough is enough on your case.

@ambiorixg12 if you see in the settings that I sent this parameter was defined as 120 in aor section qualify_frequency : 120

Thanks @jcolp I just executed a reload and started the SIP OPTIONS.

Qualify OPTIONS on Asterisk 18 with PJSIP and Realtime

Asterisk Asterisk SIP

lmoraes April 29, 2021, 1:15pm 4

Looks like SIP OPTIONS require a REGISTER first in ASTERISK 18

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Qualify OPTIONS on Asterisk 18 with PJSIP and Realtime

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