Calls from asterisk getting received 486 busy here

Asterisk Asterisk SIP
1

Hello,

In the Asterisk setup, I tested the calls with SIP endpoints. But I got 486 busy here after 180 ringing, but I initially thought this was an issue on the SIP client, but this issue occurred on all of the endpoints.
[2025-11-03 13:10:31] VERBOSE[3981070] res_pjsip_logger.c: <— Transmitting SIP request (1505 bytes) to TCP:10.70.95.86:49185 —>
INVITE sip:4019@10.70.95.86:49185;transport=TCP; SIP/2.0
Via: SIP/2.0/TCP 10.22.3.7:5060;rport;branch=z9hG4bKPj175ee753-9aea-41e6-b5c9-e4351c844ee2;alias
From: “Example 1” sip:4014@10.22.3.7;tag=81ed7506-cc54-40e2-82f3-7343ee867068
To: sip:4019@10.70.95.86
Contact: sip:asterisk@10.22.3.7:5060;transport=TCP
Call-ID: 8f487527-a49d-49d1-8da7-1aa5e123aee9
CSeq: 9666 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.0.0
Content-Type: application/sdp
Content-Length: 441

v=0
o=- 561654536 561654536 IN IP4 10.22.3.7
s=Asterisk
c=IN IP4 10.22.3.7
t=0 0
m=audio 11294 RTP/AVP 107 8 101 103
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
m=video 18558 RTP/AVP 100 99
a=rtpmap:100 VP8/90000
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=sendrecv

[2025-11-03 13:10:31] VERBOSE[3981069] res_pjsip_logger.c: <— Received SIP response (497 bytes) from TCP:10.70.95.86:49185 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.22.3.7:5060;rport;branch=z9hG4bKPj175ee753-9aea-41e6-b5c9-e4351c844ee2;alias
From: “Example” sip:4014@10.22.3.7;tag=81ed7506-cc54-40e2-82f3-7343ee867068
To: sip:4019@10.70.95.86
Call-ID: 8f487527-a49d-49d1-8da7-1aa5e123aee9
CSeq: 9666 INVITE
Content-Length: 0

[2025-11-03 13:10:31] VERBOSE[3981069] res_pjsip_logger.c: <— Received SIP response (574 bytes) from TCP:10.70.95.86:49185 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 10.22.3.7:5060;rport;branch=z9hG4bKPj175ee753-9aea-41e6-b5c9-e4351c844ee2;alias
From: “Example” sip:4014@10.22.3.7;tag=81ed7506-cc54-40e2-82f3-7343ee867068
To: sip:4019@10.70.95.86;tag=xCpmCbg
Call-ID: 8f487527-a49d-49d1-8da7-1aa5e123aee9
CSeq: 9666 INVITE
User-Agent: Unknown
Supported: replaces, outbound, gruu, path
Content-Length: 0

[2025-11-03 13:10:31] VERBOSE[3981069] res_pjsip_logger.c: <— Received SIP response (576 bytes) from TCP:10.70.95.86:49185 —>
SIP/2.0 486 Busy here
Via: SIP/2.0/TCP 10.22.3.7:5060;rport;branch=z9hG4bKPj175ee753-9aea-41e6-b5c9-e4351c844ee2;alias
From: “Example” sip:4014@10.22.3.7;tag=81ed7506-cc54-40e2-82f3-7343ee867068
To: sip:4019@10.70.95.86;tag=xCpmCbg
Call-ID: 8f487527-a49d-49d1-8da7-1aa5e123aee9
CSeq: 9666 INVITE
User-Agent: Unknown
Supported: replaces, outbound, gruu, path
Content-Length: 0

The network person is telling everything is fine with routing and VLANs. then anything missing here to get this one, but I know the call is disconnected automatically on every device. do i need to change the configurations of asterisk?

2

You’re going to need to be more specific. For example, what is this calling?

3

We are making SIP calls between two clients once the call has been recieved by the reciever and I could able to see the 180 ringing from the SIP clients in the fraction of seconds but eventually the calls are got hang up automatically and recieved 486 busy here in the asterisk server. these calls are basic SIP calls from Phone A to Phone B.

4

Does whatever is at 10.70.95.86 have do not disturb enabled?

5

Initially thought so, but the all of the endpoints having the same behaviour. so this isn’t endpoint specifc

6

Maybe they don’t like the offer of video?

Without any insight from the endpoint, you’re basically just guessing.