Hi, maybe it's an obvious 'discovery' for you, but i think i figured out how to wait for the call to be answerred when executing an outbound, mod_httapi-controlled call:
simply use the 'wait_for_answer' application before starting the playback.

Here's my current dial script:
<document type="text/freeswitch-httapi">
  <params>
    <stage>1</stage>
  </params>
  <variables />
  <work>
    <execute application="wait_for_answer" />
    <execute application="answer" />
    <pause milliseconds="1000" />
    <playback file="xyz.wav" name="user_answer" loops="2" digit-timeout="20000">
      <bind>~\d</bind>
    </playback>
  </work>
</document>
 
Of course you need to enable 'wait_for_answer' application in mod_httapi config file.
PS but if you know the answer to  to my 2nd question please post it here because I'm not really happy having to hard-code the origination_caller_id in my call scripts

Best regards
RG



On Fri, Jul 12, 2013 at 4:06 PM, Rafal Gwizdala <rafal.gwizdala@gmail.com> wrote:
Hi,

Thanks for great support so far.
I have another two questions to you:
1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. 
My originate command looks like this:

originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668 

The '6668' extension is an httapi script that plays an audio file and collects some digits.
The problem is that my httapi ivr starts the playback  even before the mobile phone is answerred.
As a result, the called person doesn't hear the beginning of the recording.
Can I somehow make my httapi script wait for the mobile to be answerred?

2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send 
Remote-Party-ID: <sip:0000000000@sip.ipfon.pl> in INVITE request that goes to the voip gateway.
Is there a way to specify a default caller id for all outgoing connections that go through a particular gateway, so I wouldn't have to specify it in originate options?

Thanks
RG