Using asterisk 13.11.2/PJSIP my calls abort after 15 minutes, when the peer sends an UPDATE message like this:
09:49:16.905139 IP (tos 0x80, ttl 251, id 0, offset 0, flags [none], proto UDP (17), length 575) 188.8.131.52.5060 > 184.108.40.206.5060: SIP, length: 547 UPDATE sip:LOCAL-NUMBER@220.127.116.11:5060 SIP/2.0 Via: SIP/2.0/UDP 18.104.22.168:5060;branch=z9hG4bKjqtq0g10dgig579c9370smp0lutp3.1 To: <sip:LOCAL-NUMBER@sip.alice-voip.de;user=phone>;tag=62df3df7-8f8e-40b5-a4e2-347c3c56f3c8 From: <sip:REMOTE-NUMBER@sip.alice-voip.de;user=phone>;tag=669730326-1476171253436 Call-ID: c670df90-4fd7-4d4c-b3ba-37bd765e56bc CSeq: 424992755 UPDATE Max-Forwards: 68 Content-Length: 0 Contact: <sip:REMOTE-NUMBER@22.214.171.124:5060;transport=udp> Supported: timer Min-SE: 120 Session-Expires: 1800;refresher=uac
As asterisk does not respond to the above UPDATE message, the peer keeps resending the UPDATE an additional 10 times, followed by a BYE message that is also resent a number of times as there is no response from asterisk either.
I understand this has something to do with session timers, but what exactly is the problem here, why does asterisk not respond to the UPDATE? Is there something I needed to configure?