Package: asterisk Version: 1:1.8.8.1-1digium1~squeeze Severity: important I've marked this important because Asterisk allows calls to continue even after the user hangs up.
I am using v1.8.8. I've also tried to use 1.8.13 (the version in wheezy) to see if that resolves the problem, but 1.8.13 has more severe problems that prevent testing of the same TLS environment: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=683956 Basically, the user agents are Polycom phones, connecting to Asterisk using TLS. A sample peer config is below. The phones can connect and register. The phones can send an SIP INVITE and start a call. However, when the phone is hung up, Asterisk doesn't receive the BYE If I stop the Asterisk process and start the `repro' SIP proxy on the same IP address and port, using the same server certificate, the phones connect to the proxy using TLS and work in the correct manner. Therefore, I don't believe the phones are at fault. I've also verified the same problem making calls from Lumicall. Also, it all works as expected if I configure the phones for UDP instead of TLS. There are various problems in the Asterisk Jira bug system regarding TLS and TCP connections not staying active to receive subsequent messages during a session. Workaround: - using repro or Kamailio SIP proxy to handle TLS clients, and relay the SIP messages to Asterisk over UDP. Only suitable if SIP packets are below the MTU (when SRTP and ICE are used, the SIP message length can exceed 1500 bytes) http://packages.debian.org/search?keywords=repro -- To UNSUBSCRIBE, email to [email protected] with a subject of "unsubscribe". Trouble? Contact [email protected]
