[Asterisk-Users] PSTN calls do NOT hang up

[Radius]

Hi all,

In my Asterisk setup, incoming calls through Cisco PSTN gateway to Asterisk extensions sounds work fine. All calls can be terminated properly after hangup. However, when calls were forwarded to voicemail, after recording & hangup the PSTN calls and cisco FXO port remained connected unless cisco port was manually shut/no shut. # key used to hang up the call did NOT help. Did anyone experience the same problem??

------------------

sip*CLI>
    -- Executing Answer("SIP/-0811b4b8", "") in new stack
    -- Executing Wait("SIP/-0811b4b8", "1") in new stack
    -- Executing VoiceMail("SIP/-0811b4b8", "u6917") in new stack
    -- Playing 'voicemail/default/6917/unavail' (language 'en')
    -- Playing 'vm-intro' (language 'en')
    -- Playing 'beep' (language 'en')
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm, 0x81254f8
    -- x=1, open writing:  /var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49, 0x80fb178
    -- x=2, open writing:  /var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav, 0x811af70
    -- Playing 'vm-msgsaved' (language 'en')
    -- Executing Hangup("SIP/-0811b4b8", "") in new stack
  == Spawn extension (sip, 6917, 4) exited non-zero on 'SIP/-0811b4b8'
sip*CLI>

---------------------------


cisco#sh voice call

1/0/1
      vtsp level 0 state = S_CONNECTvpm level 1 state = FXOLS_CONNECT vpm level 0 state = S_UP


--------------------------

dial-peer voice 999 voip
 destination-pattern 8...
 session protocol sipv2
 session target ipv4:10.1.1.1:5065
 session transport udp
 codec g711ulaw
 no vad
!
------------------------------------
exten => 6917,1,Answer
exten => 6917,2,Wait(1)
exten => 6917,3,VoiceMail(u${EXTEN})
exten => 6917,4,Hangup

Thanks.
Ben
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