[Asterisk-Users] Asterisk "no one is available to take your call"

[Howard Lowndes]

On Wed, 2005-02-16 at 11:05, Greg Oliver wrote:
> OK - I can successfully make calls from SIp phone through an asterisk 
> 323 channel to a Cisco Call Manager and out a MGCP controlled gateway.
> 
> The problem is that if the call is not answered within ~5 seconds, * 
> gives the message "no one is available to take your call" and 
> disconnects the call.  If I answer b4 the 5 seconds - everything is good.
> 
> Anywhere I need to set to get around this.
> 
> I have tried the t,T settings (even though the docs say no entry is 
> forever) with no luck.

Read the doco on the Dial command again.  It's noting to do with the Tt
option, it's the parameter before that that you need to set to the
timeout
> 
> Thanks,
> 
> Greg Oliver
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-- 
Howard.
LANNet Computing Associates;
Your Linux people <http://www.lannetlinux.com>
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