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 [EMAIL PROTECTED] 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
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