Erroneous Invite with "%40" character

I have chansip binding to port 5060 and pjsip to 5160.
Running into a strange problem when I create a new chansip endpoint and making a call to it.

The Invite the server is sending is messed up.

INVITE sip:18009220204%40Vitelity-Outbound_PJSIP@64.2.142.93 SIP/2.0

I am dialing 18009220204.
Why is the server turning this into 18009220204%40Vitelity-Outbound_PJSIP?

Vitelity-Outbound_PJSIP used to be a pjsip trunk endpoint with the same trunk host IP address, that has been deleted and does no longer exist.

Has anybody seen this before?
Asterisk 13.18.4

INVITE sip:18009220204%40Vitelity-Outbound_PJSIP@64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xx.xx:5060;branch=z9hG4bK5ad22270;rport
Max-Forwards: 70
From: <sip:8321231912@xxx.xxx.xx.xx>;tag=as77bfbd88
To: <sip:18009220204%40Vitelity-Outbound_PJSIP@64.2.142.93>
Contact: <sip:8321231912@xxx.xxx.xx.xx:5060>
Call-ID: 25f797b42aa36ce440f4a5021bc02b5a@208.105.62.60:5060
CSeq: 102 INVITE
User-Agent: FPBX-13.0.195.1(13.18.3)
Date: Fri, 25 May 2018 15:35:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 343
Chansip trunk settings:

Endpoint setting:

  [Vitelity_Out]
    host=64.2.142.93
    type=peer
    context=from-trunk

You haven’t shown the console output, such as the Dial line. It is entirely possible that the Dial line is specifying something like that resulting in it being encoded.

The dial line is indeed specifying something incorrect.
The @Vitelity-Outbound_PJSIP used to be an endpoint that no longer exists. Don’t know why it still lingers around.
Definitely some buggish behaviour, but an Asterisk restart fixed it.

-- Executing [s@macro-dialout-trunk:21] Dial("PJSIP/5315-000003ec", "SIP/Vitelity_Outbound_SBC/18009220204@Vitelity-Outbound_PJSIP,300,") in new stack