[Asterisk-biz] Asterisk Calls - Extensions (Users) do not disconnect and remaoin in Cache

[Kanuri, Seshu]

I have fired off Asterisk Extensions conf with 2 extensions i.e 2000 and 2001 and made some test calls.
I forgot to set a time out. The calls between these two were partially successful. 

I have modified the original sip.conf and extension.conf file instead of writing mine.
 
After writing my own files, it started working. I went through following steps...
1. Dialed with 2000 as user and password same as user.
2. It logged into the server.
3. When I dialed for 2001, it went to voicemail box and asked me to drop message as the 2001 was offline.
4. But when I changed username and password from the same system... its not allowing me to test the system .
5. It has cached all the entries which I unable to remove.
 
But now the problem is that these two extensions are still waiting and I do not know how to kill them.
Please guide me to remove these entries as I have tried by restarting SER and ASTERISK servers.

Can somebody help?

Seshu Kanuri


-----Original Message-----
From: asterisk-biz-admin at lists.digium.com
[mailto:asterisk-biz-admin at lists.digium.com]On Behalf Of Peter Nixon
Sent: Tuesday, June 22, 2004 3:57 AM
To: asterisk-biz at lists.digium.com
Subject: Re: [Asterisk-biz] Connect more Voip Provider to create a
network


On Friday 18 June 2004 11:40, ePyron Felix Deierlein wrote:
> Hello,
>
> > 	why we cant made a sort of network througt Voip
> > Provider from all
> >
> > > over the world to have nice price in local termination?
> > > Please say some considaration about it.
> > > Thanks in advance
> > > Dimitri
> >
> > http://www.arbinet.com/
>
> does anybody work with them? I have checked the site and it seems to be
> very intresting but also not easy...

Yes. We do. They are for high volume interconnects only..

-- 

Peter Nixon
http://www.peternixon.net/
PGP Key: http://www.peternixon.net/public.asc
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