[Asterisk-Users] BYE from Cisco gateway

[Torbjørn Lium]

Snippet from my extensions.conf

[pstn-ut]
exten => _0.,1,Dial(SIP/${EXTEN:1}@${PSTN-GW})
exten => _0.,2,Congestion
exten => _0.,3,Hangup

Probably something is
a) horribly wrong
b) easy to fix

but how?

barney wrote:
> It was also my problem...
> 
> Beware of generating ringtone (r, or rt string at the end of the call 
> command).
> 
> -b
> 
> ----- Original Message ----- From: "Torbjørn Lium" 
> <torlium at stud.aitel.hist.no>
> To: <asterisk-users at lists.digium.com>
> Sent: Tuesday, May 10, 2005 12:58 PM
> Subject: [Asterisk-Users] BYE from Cisco gateway
> 
> 
>> I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN.
>> If a user on a softphone hangs up first the PSTN port on the cisco is 
>> released and new calls can be made on the same voice port. But when 
>> the user on the PSTN side hangs up first the voice port on the cisco 
>> stays open until the user on the softphone hangs up.
>> Any ideas what I'm doing wrong?
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> 
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