[Asterisk-Users] PSTN calls do NOT hang up

[Stig Andersson]

Hi,

Asterisk either need to know when the remote caller ends his call,
or it must detect the silence. 

Simplest solution is to activate silence detection, see voicemail.conf.
You may need to do some testing to get the proper "silencethreshold" setting. 

Also search the archive, this is a often discussed issue...
http://mharc.lists.openservices.ca/archives/html/asterisk-users/

/Stig


At 17:25 2004-04-07 +0800, you wrote: 
>
> 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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