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<DIV><FONT face=Arial size=2>Hi all,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>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??</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>------------------</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>sip*CLI></FONT></DIV>
<DIV><FONT face=Arial size=2> -- Executing
Answer("SIP/-0811b4b8", "") in new stack<BR> -- Executing
Wait("SIP/-0811b4b8", "1") in new stack<BR> -- Executing
VoiceMail("SIP/-0811b4b8", "u6917") in new stack<BR> --
Playing 'voicemail/default/6917/unavail' (language 'en')<BR>
-- Playing 'vm-intro' (language 'en')<BR> -- Playing 'beep'
(language 'en')<BR> -- x=0, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: gsm,
0x81254f8<BR> -- x=1, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav49,
0x80fb178<BR> -- x=2, open writing:
/var/spool/asterisk/voicemail/default/6917/INBOX/msg0003 format: wav,
0x811af70<BR> -- Playing 'vm-msgsaved' (language
'en')<BR> -- Executing Hangup("SIP/-0811b4b8", "") in new
stack<BR> == Spawn extension (sip, 6917, 4) exited non-zero on
'SIP/-0811b4b8'<BR></FONT><FONT face=Arial size=2>sip*CLI></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>---------------------------</DIV>
<DIV><BR></DIV>
<DIV><FONT face=Arial size=2>cisco#sh voice call<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>1/0/1<BR> vtsp level
0 state = S_CONNECTvpm level 1 state = FXOLS_CONNECT vpm level 0 state =
S_UP<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV></FONT>
<DIV><FONT face=Arial size=2>--------------------------</FONT></DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT>
<DIV><FONT face=Arial size=2>dial-peer voice 999
voip<BR> destination-pattern 8...<BR> session protocol
sipv2<BR> session target ipv4:10.1.1.1:5065<BR> session transport
udp<BR> codec g711ulaw<BR> no vad<BR>!</FONT></DIV>
<DIV><FONT face=Arial size=2>------------------------------------<BR>exten =>
6917,1,Answer<BR>exten => 6917,2,Wait(1)<BR>exten =>
6917,3,VoiceMail(u${EXTEN})<BR>exten => 6917,4,Hangup<BR></FONT></DIV>
<DIV><FONT face=Arial size=2>Thanks.</DIV></FONT>
<DIV><FONT face=Arial size=2>Ben</FONT></DIV></BODY></HTML>