[asterisk-users] Freepbx VPN SIP Client (SIP/2.0 401 Unauthorized)

basti mailinglist at unix-solution.de
Fri Nov 6 07:28:18 CST 2020

i try to connect my SIP Client (linphone) via VPN to FreePBX.
The routing looks OK. I can ping the Endpoints and traffic is routing.
I can also Register my Sip Client.

debpbx*CLI> pjsip list contacts

   Contact:  <Aor/ContactUri..............................> <Hash....> 
<Status> <RTT(ms)..>

   Contact:  731/sip:731 at                163a967d99 
Avail        15.722
   Contact:  734/sip:734 at                    1b1aa8cbac 
Avail        62.180

So far so good. When I try to an other extension I get a timeout.

root at debpbx:/etc/asterisk# tcpdump -ni enp0s15 host and not 
port 80
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on enp0s15, link-type EN10MB (Ethernet), capture size 262144 bytes
13:03:04.086687 IP > SIP: INVITE 
sip:731 at asterisk.kes SIP/2.0
13:03:04.087364 IP > SIP: SIP/2.0 
401 Unauthorized
13:03:04.126101 IP > SIP: ACK 
sip:731 at asterisk.kes SIP/2.0
13:03:09.054643 IP > SIP
13:03:14.112561 IP > SIP: OPTIONS 
sip:734 at SIP/2.0
13:03:14.162609 IP > SIP: SIP/2.0 200 Ok
13:03:19.057752 IP > SIP
13:03:29.060765 IP > SIP
13:03:44.672509 IP > SIP

I think the SIP/2.0 401 Unauthorized is the problem.
I also had add the VPN IP range to the local_net but that does not solve 
the problem.

root at debpbx:/etc/asterisk# grep -ri 10.8.0

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