[asterisk-users] calls don't hang up correctly on VM

Danny Nicholas danny at debsinc.com
Fri Jul 23 08:35:01 CDT 2010


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 


More information about the asterisk-users mailing list