[Asterisk-Users] Dial to a SIP fone ends up at Voicemail Busy

[Terry H. Gilsenan]

Hi,

I was having this problem with Gradstream BT101's with Asterisk @ Home
version 0.7.

The problem was that there was a sip channel still open (as far as asterisk
and the phone were concerned) however this sip channel was not actually in
use. The existence of this sip channel meant that whilst the phone could
make calls, any incoming calls were directed to voicemail.

At the time what we had to to was to kill those persistent sip channels.

I have not upgraded to Asterisk @ home version 1.0 and the problem no longer
occurs.

What version of Asterisk are you using?

Connect to the CLI and issue the commands

 sip show channels

 zap show channels

Are there any channels showing there that you know should be down?

Regards,
T


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Arnd Vehling
> Sent: Wednesday, 25 May 2005 6:28 PM
> To: C F; Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Dial to a SIP fone ends up at 
> Voicemail Busy
> 
> C F wrote:
> 
> > are these phones behind nat?
> 
> Yes, but correctly registered. The same fones dont have any 
> problems when registered to a SER Server.
> 
> Can constantly reloading the configuration cause problems?
> 
> cheers,
> 
>     Arnd
> 
> 
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