tayler6000
Calls hang up after 30 seconds.
This happens on both DEV and MASTER branch
CSeq: {'check': '102', 'method': 'BYE'} User-Agent: voicecall Proxy-Authorization: Digest username="rah", realm="x.voip.ms", algorithm=MD5, uri="sip:x.voip.ms", nonce="086055f5", response="fa02cb023c08d6aa4f101b7179e4ae0e" X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18 Content-Length: 0
Body:
Raw: b'BYE sip:[email protected]:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 208.100.60.8:5060;branch=z9hG4bK39610d4c;rport\r\nMax-Forwards: 70\r\nFrom: sip:x.voip.ms;tag=as311a06ae\r\nTo: sip:[email protected];tag=db29078c\r\nCall-ID: [email protected]:5060\r\nCSeq: 102 BYE\r\nUser-Agent: voicecall\r\nProxy-Authorization: Digest username="281267_rah", realm="chicago.voip.ms", algorithm=MD5, uri="sip:chicago1.voip.ms", nonce="086055f5", response="fa02cb023c08d6aa4f101b7179e4ae0e"\r\nX-Asterisk-HangupCause: No user responding\r\nX-Asterisk-HangupCauseCode: 18\r\nContent-Length: 0\r\n\r\n' BYE recieved
Using dev branch
I believe some sort of heartbeat is not being replied to. Found this article that has info about the specific error message/hangup cause https://community.asterisk.org/t/solved-incoming-calls-hangup-after-30-seconds/42274/4
Caught this issue too. In my example there are Ubuntu with Asterisk for two numbers (1001 and 1002) on IP 192.168.1.100 and two Win10 machines on 192.168.1.1-2 with phones on it. Both register normally (but for some reason re-register after couple of minutes) but when i make a call it is then cut after 30 to 32 seconds from answer. At the last moment before the disaster SERVER for no reason sends SIP INVITE for both of the host (actually this moment vary, sometimes it sends it to only one of them idk why). Then few last RTP packets are sent (or not, varies as SIP INVITE) and then 10 or so BYE sent from server to each host with ACKs in rensponse.
