Hello, I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM
URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP
servers (it doesn’t recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I
have highlighted the issue in Yellow. Does anyone know of a solution / workaround for this issue?

<--- Received SIP request (648 bytes) from UDP:172.16.60.160:5061 --->
INVITE sip:039988120F@172.16.60.160:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK-29450-3-0
Max-Forwards: 60
From: ;tag=3
To:
Call-ID: 3-29450@172.16.60.160
CSeq: 1 INVITE
Contact:
User-Agent: Simulator Supported: 100rel Privacy: id Min-SE: 90
Content-Type: application/sdp Content-Length: 201

v=0
o2.16.60.160 10864 2 IN IP4 172.16.60.160
s=SIP Call c=IN IP4 172.16.60.160
t=0 0
a=sendrecv m=audio 60000 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<--- Transmitting SIP response (730 bytes) to UDP:172.16.60.160:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060
;rport;received2.16.60.160;branch=z9hG4bK-29450-3-0
Call-ID: 3-29450@172.16.60.160
From: ;tag=3
To: ;tagO7ef94f-fb15-4bf5-94bd-4e43fe-299655
CSeq: 1 INVITE
Contact:
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE, REGISTER
Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 189

v=0
o=- 10864 4 IN IP4 10.2.0.67
s=Asterisk c=IN IP4 172.16.60.160
t=0 0
m=audio 19404 RTP/AVP 8
c=IN IP4 172.16.60.160
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:150
a=sendrecv