Dialogue between pbx and cisco VG224

I just enabled pjsip logger and noted a big SIP traffic between a cisco VG224 and the pbx.
Each of 24 extensions registered from VG224 into freepbx generates continously a dialogue like this :

<— Transmitting SIP request (440 bytes) to UDP:192.168.200.213:53930 —>
OPTIONS sip:[email protected]:53930 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.200:5060;rport;branch=z9hG4bKPj6a4536df-fda1-4949-a5e0-4d60d338a9b0
From: sip:[email protected];tag=c66fbbe8-b96f-44dd-9d64-8c81e700bcb4
To: sip:[email protected]
Contact: sip:[email protected]:5060
Call-ID: 0765133f-e603-4304-989b-1033ff40805f
CSeq: 54669 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-16.0.40.7(18.17.1)
Content-Length: 0

<— Received SIP response (823 bytes) from UDP:192.168.200.213:53930 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.200:5060;rport;branch=z9hG4bKPj6a4536df-fda1-4949-a5e0-4d60d338a9b0
From: sip:[email protected];tag=c66fbbe8-b96f-44dd-9d64-8c81e700bcb4
To: sip:[email protected];tag=EE6203B4-22A
Date: Mon, 30 Oct 2023 16:24:54 GMT
Call-ID: 0765133f-e603-4304-989b-1033ff40805f
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 54669 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 175

v=0
o=CiscoSystemsSIP-GW-UserAgent 694 2267 IN IP4 192.168.200.213
s=SIP Call
c=IN IP4 192.168.200.213
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 192.168.200.213

Where 192.168.200.200 is pbx and 192.168.200.213 is Cisco VG224

Is this traffic OK, or something anomalous to be worried of ?

Thank you

That traffic is fine. Asterisk is sending an OPTIONS to ensure the remote side is reachable, and it’s getting back a response.

Good, thank you !

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