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

[Greg Oliver]

True, but it also states that with no timeout value that it will dial 
until the caller hangs up.

I have included my dial pattern - can anyone see anything that would 
cause this, or something in my sip.conf or h323.conf files that would 
override these settings?

Thanks,

Greg Oliver

[outbound]
exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} )
exten => _4XXX,2,Dial(H323/${EXTEN})
exten => _4XXX,3,Congestion

exten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) exten => 
_5XXX,2,Dial(H323/${EXTEN})
exten => _5XXX,3,Congestion

exten => _9NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _9NXXXXXXXXX,2,Dial(H323/${EXTEN})

exten => _91NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN})
exten => _91NXXXXXXXXX,2,Dial(H323/${EXTEN})


default context includes outbound, and contexts in sip.conf and 
h323.conf are using default.  Like I say, call answered before ~5 
seconds are fine, other than that it is transferred to 4607..

Howard Lowndes wrote:
> 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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