[Asterisk-Users] Using call.sample on Zap hardware - Answering problem

[Patrick Healy]

Hi,

 

First let me apologize if you've seen this question before recently. I
registered using an address that had a Lotus Notes e-mail client and all
messages to the list ended up being unreadable. Love that lotus notes.

 

Anyway,  to the problem -

 

I've got a X100P connected to a POTS line and am using it to call out to
play a recorded message.  I drop a copy of sample.call into
/var/spool/asterisk/outgoing and Asterisk picks up the line and initiates
the call.  The problem is that the recorded message starts immediately and
doesn't wait for the called party to pick up the phone.  When I try this
same process with a SIP extension, the process works like a champ, it just
fails on the Zap interface.

 

Is there some kind of setting or adjustment that I can make to the Zap
configuration that will allow it to  wait until the phone is answered?

 

Here's the relevant portion of extensions.conf for that entry.

 

 

[outgoing]

exten => s,1,DigitTimeout,5

exten => s,2,ResponseTimeout,10

exten => s,3,Wait(4)

exten => s,4,Answer

exten => s,5,Background(demo-congrats)            ;           Play some
recordings for testing purposes only.

exten => s,6,Background(demo-instruct)

exten => 1,1,Goto(s,5)  

exten => 2,1,Goto(msgack,s,1) 

exten => t,1,Playback(vm-goodbye)

exten => t,2,Hangup

 

[msgack]

exten => s,1,Playback(auth-thankyou)

exten => s,2,Playback(vm-goodbye)

exten => s,3,Hangup

 

 

Thanks!

 

Pat Healy

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