Simple HA architecture for Asterisk

Asterisk Asterisk Support
1

I have a simple HA architecture configured for VoIP. On server #1, Asterisk + FreePBX are running. On the second server, Asterisk + FreePBX are also installed, but these services are only started when server #1 is inactive. Both servers use a database cluster and share the same database. Configuration between the servers is fully synchronized. I’ve tested it, and the solution works well.

Both Asterisk servers use a VIP address. Phones connect to the Asterisk instance on the first server correctly and everything works fine. However, after switching to the failover server, I see that the phones connect and register (they also appear as registered on the Asterisk side). However, when I try to make a call, the phone displays the message “Service not available.”

I’m using Ubuntu 24.04 LTS, Asterisk 20.6 (from Ubuntu packages), and FreePBX 17.

My question is: why is this happening? Besides configuration, is there anything else that needs to be synchronized?

2

On Monday 13 October 2025 at 22:41:44, autominus via Asterisk Community wrote:

I have a simple HA architecture configured for VoIP.

after switching to the failover server, I see that the phones connect and
register (they also appear as registered on the Asterisk side). However,
when I try to make a call, the phone displays the message “Service not
available.”

Show us a SIP packet capture from the failover server of such a call, and also
the output of “pjsip show endpoints” from the same server at that time.

Antony.

–
“It wouldn’t be a good idea to talk about him behind his back in front of
him.”

  • murble
3

When an IP floats from one box to the other, make sure ARP tables are updated appropriately. Without this, IP phones may keep sending traffic to the previous box.

4

After switching to failover, I see the following in the communication:

Request: INVITE sip:9910@voip.voipsystems.lan |
Status: 100 Trying |
Status: 503 Service Unavailable | 
Request: ACK sip:9910@vip.voipsystems.lan |

On the phone, the message “Service Unavailable” is displayed.

I checked and I can see that all the phones are registered.


 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================

 Endpoint:  9910/9910                                            Not in use    0 of inf
     InAuth:  9910-auth/9910
        Aor:  9910                                               1
      Contact:  9910/sip:9910@10.200.5.53:5060             5d17dad8ad Avail        10.482

 Endpoint:  9920/9920                                            Not in use    0 of inf
     InAuth:  9920-auth/9920
        Aor:  9920                                               1
      Contact:  9920/sip:9920@10.200.5.38:5060             625740274b Avail        10.667
5

The communication log file [failover] - VoIP calls not working.

The communication log file [promary] - VoIP calls working.

6

When capturing traffic on the interface that previously hosted the floating IP, can you see the above INVITE message coming in ?

7

INVITE messages are being delivered correctly — that is to the server that currently has the floating IP address assigned.

8

While troubleshooting and enabling pjsip set logger on, I found the source of the problem:

– Executing [s@macro-dial-one:27] GosubIf(“PJSIP/9910-00000004”, “1?dstring,1():dlocal,1()”) in new stack
– Executing [dstring@macro-dial-one:1] Set(“PJSIP/9910-00000004”, “DSTRING=”) in new stack
– Executing [s@macro-dial-one:63] NoOp(“PJSIP/9910-00000004”, “Returned from dial-one with nothing to call and DIALSTATUS: NOANSWER”).

Digging deeper, after running the command database show AMPUSER/9910/device, I see:

0 results found

On the failover server, there’s an inconsistency between the astdb.sqlite3 files — the one on the primary server is larger in size. It’s clear that the database is missing the necessary entries.

When I copied the astdb.sqlite3 file from the primary server to the failover server, the calls started working. However, I think that copying or synchronizing this file is not the best idea. I have the impression that this file should be rebuilt during the startup of Asterisk and FreePBX.

Any ideas?