Hello List, I'm moving my asterisk testing installation from CENTOS 5.4 on a real machine to SUSE on a xen VM. Everything seemed to go off without a hitch until I really looked at it. The call answers and processes correctly, but when it is time to end the call, the phone never disconnects from asterisk. For a simple functionality test, I use this "Monkeys" snippet to tell me if all is well: dialplan show 99 at default [ Context 'default' created by 'pbx_config' ] '99' => 1. Answer() [pbx_config] 2. Playback(tt-monkeys) [pbx_config] 3. Noop(tt-monkeys) [pbx_config] 4. Hangup() [pbx_config] -= 1 extension (4 priorities) in 1 context. =- This is my CLI output from a test call (1.6.2.9) *CLI> == Using SIP RTP CoS mark 5 -- Executing [99 at default:1] Answer("SIP/170-00000000", "") in new stack -- Executing [99 at default:2] Playback("SIP/170-00000000", "tt-monkeys") in new stack -- <SIP/170-00000000> Playing 'tt-monkeys.gsm' (language 'en') -- Executing [99 at default:3] NoOp("SIP/170-00000000", "tt-monkeys") in new stack -- Executing [99 at default:4] Hangup("SIP/170-00000000", "") in new stack == Spawn extension (default, 99, 4) exited non-zero on 'SIP/170-00000000' -- Executing [h at default:1] Set("SIP/170-00000000", "CDR(userfield)Hangupcause:16") in new stack -- Executing [h at default:2] Verbose("SIP/170-00000000", "details - time time2 status ") in new stack details - time time2 status -- Executing [h at default:3] GotoIf("SIP/170-00000000", "0?end-call,s,1") in new stack -- Executing [h at default:4] Verbose("SIP/170-00000000", "details - time time2 status ") in new stack details - time time2 status -- Executing [h at default:5] NoOp("SIP/170-00000000", "id 1279891796.0 time 16") in new stack -- Executing [h at default:6] NoOp("SIP/170-00000000", "caller hung up eh") in new stack -- Executing [h at default:7] Goto("SIP/170-00000000", "end-call,s,1") in new stack -- Goto (end-call,s,1) -- Executing [s at end-call:1] NoOp("SIP/170-00000000", "Verbose(details - time time2 status )") in new stack -- Executing [s at end-call:2] Hangup("SIP/170-00000000", "") in new stack == Spawn extension (end-call, s, 2) exited non-zero on 'SIP/170-00000000' The call executed as expected, but the telephone never hangs up If I do an IAX call, the hangup occurs as expected. Any clues? Thanks Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100723/81f90cf3/attachment.htm