I recently setup an ATA equipament in my Asterisk 13 PJSIP behind NAT, and this equipament is lost the connection after the default_expiration time that was 2 minutes.
I was check it and observed that this equipment is not answering for SIP OPTIONS monitoring, I don’t know why because when the equipment register is sent a packet of SIP OPTIONS then the endpoint answer 200 OK, then in the next packets none answer is received.
First SIP OPTION = answer 200 ok
xOPTIONS sip:16028019@177.74.x.y:29156 SIP/2.0
172.31.13.68:5060 177.74.x.y:29156 xVia: SIP/2.0/UDP z.h.162.y:5060;rport;branch=z9hG4bKPjb1c7a8fb-3554-47a2-bcb1-4cdd7e779809
qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqq xFrom: <sip:16028019@172.31.13.68>;tag=94086be2-fc7d-454e-a482-63baf9330198
x OPTIONS x xTo: <sip:16028019@177.74.x.y>
12:23:02.422342 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x xContact: <sip:16028019@z.h.162.y:5060>
+0.013616 x 200 OK x xCall-ID: e5a3820c-2d90-478a-9e84-fbe4f5629bdb
12:23:02.435958 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x xCSeq: 4910 OPTIONS
x x xMax-Forwards: 70
x x xUser-Agent: Asterisk PBX 13.21.1
x x xContent-Length: 0
Second attempt to SIP OPTION = no answer
xOPTIONS sip:16028019@177.74.x.y:11671 SIP/2.0
172.31.13.68:5060 177.74.x.y:11671 xVia: SIP/2.0/UDP z.h.162.y:5060;rport;branch=z9hG4bKPj6f7f40f7-effb-4108-9816-984a85e52183
qqqqqqqqqqwqqqqqqqqq qqqqqqqqqqwqqqqqqqqq xFrom: <sip:16028019@172.31.13.68>;tag=4820c6e0-3e93-4f8a-8196-29662d6c2d8e
x OPTIONS x xTo: <sip:16028019@177.74.x.y>
12:23:25.223202 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x xContact: <sip:16028019@z.h.162.y:5060>
+1.000578 x OPTIONS x xCall-ID: 562c3a08-1fad-4ab4-9eb3-92c952b44a52
12:23:26.223780 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x xCSeq: 34424 OPTIONS
+1.999601 x OPTIONS x xMax-Forwards: 70
12:23:28.223381 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x xUser-Agent: Asterisk PBX 13.21.1
+4.000632 x OPTIONS x xContent-Length: 0
Anyone used that and worked well? Is it possible I disable the SIP option for PJSIP?
In the same internal network we have other VoIP telephones working well.