[Asterisk-Users] CRITICAL PROBLEM

[Kevin Bockman]

Tim McKee wrote:
> I'm running a large number (125) remote sip phones for FEMA in the Gulf area
> over satellite.  I've run into a major problem and need some assistance.
> 
> When dialing the FEMA voice response system, it appears that it never
> actually answers the phone.  I never get audio when dialing via SIP through
> a provider and when dialing over my PRI it actually times out with a phone
> not ansered message, though the audio is passed.  Apparently the FEMA system
> does not issue an 'ANSWERED' or 'CONNECTED' code back to the PSTN as it
> should.  The link stays in an in-progress state until timeout occurs or the
> user hangs up.
> 
> Is there any way to get SIP to pass audio prior to getting a call complete
> message?  This is Asterisk CVS-HEAD 08-01-2004.
> 
> Please respond to tmckee at sdnglobal.com as I don't have good access to the
> email account serving this list during the day.

I think for this problem, you have to do an Answer first.  You can make 
a separate extension before your other match and do an Answer before the 
Dial.

Something like this but with  Dial and change the exact match to do the 
Answer first instead of the fuzzy match:

exten => 11111111111,1,Dial(SIP/1|20)
exten => 11111111111,n,Hangup
exten => 12222222222,1,Dial(SIP/2|20)
exten => 12222222222,n,Hangup

exten => _1XXXXXXXXXX,1,Answer
etc etc etc..

Kevin