ACK sip:110@IP_PUBLIC_IP:5066 SIP/2.0

Via: SIP/2.0/UDP 172.26.101.10:5080;branch=z9hG4bK0a330ae6

Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on>

Max-Forwards: 70

From: "asterisk" <sip:110@172.26.101.10:5080>;tag=as14d7523e

To: <sip:110@172.26.101.50:8002>;tag=1749303708

Contact: <sip:110@172.26.101.10:5080>

Call-ID: 7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080

CSeq: 102 ACK

User-Agent: ast01

Content-Length: 0

With code i posted before i have now issue to answer calls from ast (generated by asterisk)

200 From Phone arrives fine to Kamailio

<--- SIP read from UDP:172.26.101.50:8002 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.26.101.10:5080;rport=5080;branch=z9hG4bK125b3b98

Record-Route: <sip:192.168.0.170:8002;nat=no;ftag=as14d7523e;lr=on>

From: "asterisk" <sip:110@172.26.101.10:5080>;tag=as14d7523e

To: <sip:110@172.26.101.50:8002>;tag=1749303708

Call-ID: 7e723f1c64086e964df79e493350a2a4@172.26.101.10:5080

CSeq: 102 INVITE

Contact: <sip:110@PUBLIC.IP:5066>

Supported: replaces, path, timer, eventlist

User-Agent: Grandstream GXV3275 1.0.3.37

Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE

Content-Type: application/sdp

Content-Length: 255

v=0

o=110 8003 8000 IN IP4 172.26.101.41

s=SIP Call

c=IN IP4 172.26.101.41

t=0 0

m=audio 8424 RTP/AVP 0 8 101

a=sendrecv

a=rtpmap:0 PCMU/8000

a=ptime:20

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sdp_proxied:yes

<------------->

I see 200 OK sent to Asterisk

Asterisk sent to me ACK but Kamailio seems to do not send to Phone behind nat

3.- 192.168.0.170:8002

Thanks