<div dir="ltr">Hi,<div><br></div><div>i am wondering what will happen if you send a SIP REFER (transfer) after validating leg-a without actually answering the call; will it still cost you 2 * 1.44</div><div><br></div><div>
:/</div><div><br></div><div>-mustafa</div><div><br><div class="gmail_quote">On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi <span dir="ltr"><<a href="mailto:mthakershi@gmail.com">mthakershi@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I am currently looking at Vitelity. They have 1.44 cents per minute. They do charge for incoming/outgoing both.<div><br>
</div><div>I am not literally dialing out to a phone. I want to dial a mobile number or any other US number for that matter.</div>
<div><br></div><div>Here is what I intent to do:</div><div>1. A US phone dials to Vitelity number -- comes to my FS box</div><div>2. I validate few things</div><div>3. Dial another US number and connect received call to that one</div>
<div><br></div><div>So as you said I will be charged 2 * 1.44 (since I can't terminate the arrived call after validation).</div><div><br></div><div>Is there any other way to the sequence I have specified above?</div>
<div>
<br></div><div>Are there providers similar to Vitelity but cheaper (with relatively same features)?</div><div><br></div><div>Thank you.<br><div><br><div class="gmail_quote">On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove <span dir="ltr"><<a href="mailto:acosgrov@gmail.com" target="_blank">acosgrov@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote:<br>
> Hello,<br>
><br>
><br>
> It would be a great help if someone can guide me.<br>
><br>
><br>
> 1. I would like to first receive a call, perform certain validations.<br>
> (Able to do this via mod_managed application that handles call from<br>
> dialplan).<br>
><br>
<br>
</div>That should not be a problem, I don't know your requirements so can't<br>
provide a full answer.<br>
<div><br>
><br>
> 2. Now, I would like to dial out to a PSTN number so that received<br>
> call is connected to this new outbound number.<br>
><br>
<br>
</div>This can be done and is called hairpinning.<br>
<div><br>
><br>
> How can this be done? Do I use Originate from within my .NET<br>
> (mod_managed) code?<br>
><br>
<br>
</div>Yes, you would be bridging the two legs like any normal call. Instead of<br>
going to an endpoint you're going back out over the PSTN.<br>
<div><br>
><br>
> Do I get charged for both incoming and outbound call until the entire<br>
> session ends? Is there a way to receive call, validate and then sort<br>
> of transfer and then terminate the received call so I do not get<br>
> charged for both?<br>
<br>
</div>That would depend on your provider but most likely yes. As for<br>
terminating one end after validation that is not going to happen. A leg<br>
terminates to you on an agreed fee schedule. No carrier that I know of<br>
supports that. Now if you kept everything SIP... you could do a transfer<br>
after the validation.<br>
<br>
<br>
<br>
Anthony C.<br>
<div><div></div><div><br>
<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Ghulam Mustafa<br>cell: +92 333.611.7681<br>sip: <a href="mailto:cyrenity@ekiga.net">cyrenity@ekiga.net</a><br>mail: <a href="http://mustafa.pk">mustafa.pk</a>@<a href="http://gmail.com">gmail.com</a><br>
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