<div dir="ltr">Hi,<div style><br></div><div style>Thanks for great support so far.</div><div style>I have another two questions to you:</div><div style>1. I'm originating a call to a mobile phone where the second leg is a httapi-based IVR service. </div>
<div style>My originate command looks like this:</div><div style><br></div><div style>originate {origination_caller_id_number=2234567890}sofia/gateway/ipfon/606999888 6668 <br></div><div style><br></div><div style>The '6668' extension is an httapi script that plays an audio file and collects some digits.</div>
<div style>The problem is that my httapi ivr starts the playback even before the mobile phone is answerred.</div><div style>As a result, the called person doesn't hear the beginning of the recording.</div><div style>
Can I somehow make my httapi script wait for the mobile to be answerred?</div><div style><br></div><div style>2. If origination_caller_id_number is not specified in the originate call FreeSWITCH will send </div><div style>
<span style="font-family:arial,sans-serif;font-size:12.800000190734863px">Remote-Party-ID: <</span><a href="mailto:sip%3A0000000000@sip.ipfon.pl" target="_blank" style="font-family:arial,sans-serif;font-size:12.800000190734863px">sip:0000000000@sip.ipfon.pl</a><span style="font-family:arial,sans-serif;font-size:12.800000190734863px">></span> in INVITE request that goes to the voip gateway.</div>
<div style>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?</div><div style><br></div><div style>Thanks</div>
<div style>RG</div></div>