After a minute or two, cannot ring device

[fejj1]

The devices are Siemens Openstage. When connected on the same network to an asterisk server (version 16), they work fine. When behind a NAT, connect fine to a telecom provider, also work fine.

But when I connect to an asterisk server (version 16) on a virtual server on the internet not behind a NAT (but device still is), works fine for a minute or two, and then any calls to the device don’t ring.

[office-phone](!)
type=endpoint
context=LocalSets

[Siemens-Openstage](!,office-phone)
disallow=all
allow=alaw
allow=ulaw
allow=g729

rtp_symmetric=yes
direct_media=no
rewrite_contact=yes

[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
max_contacts=1
remove_existing=yes


...


[200serie](!,Siemens-Openstage)
; config for the series here
callerid=<31431111111>


[Set203](200serie)
auth=auth203
aors=Set203

[auth203](auth-userpass)
password=nunubiznz
username=Set203

[Set203](aor-single-reg)

Thanks in advance!

[jcolp]

You haven’t configured a qualify_frequency, so Asterisk is not sending any OPTIONS request. This means that the NAT Mapping can close resulting in calls to the device to fail. The qualify_frequency option is set on the AOR[1]

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_pjsip#Asterisk18Configuration_res_pjsip-contact_qualify_frequency

[david551]

Also you have symmetric media, and rewrite contact, but not the other NAT hack, force rport. That actually tends to suggest that the first two aren’t needed either.

[jcolp]

The “force_rport” option defaults to “yes”, so it’s only needed to be specified if you want to disable it (ala Cisco phones).

[fejj1]

Thank you so very much! That was it.