Call Disrupted…due To Registration Of Third Server?
We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to
10.0.0.228. But sometimes another of our servers becomes listed as a SIP
agent, even though the server’s IP address isn’t part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as a consequence Asterisk re-locked on the source and destination servers…but also registered third server
10.0.0.125. This seems to have broken the call to the desired destination server.
[2020-01-14 18:08:25] VERBOSE[29350][C-00000006] res_rtp_asterisk.c:
0x7f40240322e0 — Strict RTP switching source address to 10.0.0.228:42150
[2020-01-14 18:08:26] VERBOSE[29324][C-00000006] res_rtp_asterisk.c:
0x7f403c00c3b0 — Strict RTP learning complete – Locking on source address
10.0.0.192:22522
[2020-01-14 18:08:26] VERBOSE[29350][C-00000006] res_rtp_asterisk.c:
0x7f40240322e0 — Strict RTP learning complete – Locking on source address
10.0.0.228:42150
[2020-01-14 18:09:01] VERBOSE[1389] asterisk.c: Remote UNIX connection
[2020-01-14 18:09:01] VERBOSE[29363] asterisk.c: Remote UNIX connection disconnected
[2020-01-14 18:09:47] VERBOSE[1429] chan_sip.c: Registered SIP ‘1000’ at
10.0.0.125:5060
[2020-01-14 18:09:47] VERBOSE[1429] chan_sip.c: Saved useragent
“FreeSWITCH-mod_sofia/1.6.20~64bit” for peer 1000
Is my description accurate for this log snippet?
How can we prevent the registration of third servers?
One thought on - Call Disrupted…due To Registration Of Third Server?
Asterisk 17 is the most recent major version, but I don’t know if an upgrade will solve your problem.
Is user/account 1000 sharing the same password across all 3 systems ?
After changing passwords, you might try adding some iptables rules on
.228 & .192 to reject SIP traffic from .125
Regards,
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