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don't know vitelity,<br>
but voip.ms. supports user/password or ip based auth. has built in
Vmail, redirect on network outage.<br>
us did's are .99/c per month, and calls 1.1c/ month. (20? trunks per
did?)<br>
<br>
<br>
<br>
<br>
On 8/25/10 2:28 PM, Malay Thakershi wrote:
<blockquote
cite="mid:AANLkTin85NQsuRwOOtD0gbi0k7qxXKRKrBreuSW6dUWC@mail.gmail.com"
type="cite">Can someone suggest me wholesale and cheaper provider than
Vitelity but offering similar service?
<div><br>
</div>
<div>In/Out bound both USA.</div>
<div><br>
</div>
<div>Thank you.<br>
<br>
<div class="gmail_quote">On Wed, Aug 25, 2010 at 3:33 AM, David
Ponzone <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:david.ponzone@ipeva.fr">david.ponzone@ipeva.fr</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div style="word-wrap: break-word;">AFAIK, most (all ?) providers
wont accept the transfer.
<div>They are not allowed to change the rate of the call from the
caller perspective after the call was initiated.</div>
<div>Imagine you transfer the call to an Iridium number...</div>
<div>
<div class="im"><br>
<div> <span
style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 14px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; text-indent: 0px; text-transform: none; white-space: normal; word-spacing: 0px;">
<div style="word-wrap: break-word;"><span
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<div style="word-wrap: break-word;">
<div><font face="'Helvetica Neue'"><font color="#1c00ff">David
Ponzone </font><font size="3" color="#000000"><span
style="font-size: 12px;">Direction Technique</span></font></font></div>
<div><font face="'Helvetica Neue'"><font size="3"><span
style="font-size: 13px;">email: <a moz-do-not-send="true"
href="mailto:david.ponzone@ipeva.fr" target="_blank">david.ponzone@ipeva.fr</a></span></font></font></div>
<div><font face="'Helvetica Neue'"><font size="3"><span
style="font-size: 13px;">tel: 01 74 03 18 97</span></font></font></div>
<div><font face="'Helvetica Neue'"><font size="3"><span
style="font-size: 13px;">gsm: 06 66 98 76 34</span></font></font></div>
<div><font face="'Helvetica Neue'"><br>
</font></div>
<div><font face="'Helvetica Neue'" color="#1c00ff">Service Client<span> </span></font><font
face="'Helvetica Neue'"><font color="#ff0000">IP</font></font><font
face="'Helvetica Neue'" color="#1c00ff">eva</font></div>
<div><font face="'Helvetica Neue'" color="#1c00ff"><span
style="color: rgb(0, 0, 0); font-family: Helvetica;">
<div><font face="'Helvetica Neue'"><font size="3"><span
style="font-size: 13px;">tel: 0811 46 26 26</span></font></font></div>
<div><font face="'Helvetica Neue'" size="3"><span
style="font-size: 13px;">
<div style="margin: 0px; color: rgb(0, 34, 243);"><span
style="text-decoration: underline;"><a moz-do-not-send="true">www.ipeva.fr</a></span><span
style="color: rgb(101, 104, 149);"> - <span
style="color: rgb(0, 34, 243); text-decoration: underline;"><a
moz-do-not-send="true">www.ipeva-studio.com</a></span></span></div>
<div style="margin: 0px; color: rgb(0, 34, 243);"><span
style="text-decoration: underline;"><br>
</span></div>
<div style="margin: 0px; color: rgb(0, 34, 243);"><span>
<div
style="margin: 0px; text-align: justify; color: rgb(192, 192, 192);"><i>Ce
message et toutes les pièces jointes sont confidentiels et établis à
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color="#c0c0c0"><i><br>
</i></font></div>
</span></div>
</span></font></div>
</span></font></div>
</div>
</span><br>
</div>
</span><br>
</div>
<br>
</div>
<div>
<div>Le 25/08/2010 à 07:44, Ghulam Mustafa a écrit :</div>
<div>
<div class="h5"><br>
<blockquote type="cite">
<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 moz-do-not-send="true"
href="mailto:mthakershi@gmail.com" target="_blank">mthakershi@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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 moz-do-not-send="true"
href="mailto:acosgrov@gmail.com" target="_blank">acosgrov@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; 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><br>
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<br>
-- <br>
Ghulam Mustafa<br>
cell: +92 333.611.7681<br>
sip: <a moz-do-not-send="true" href="mailto:cyrenity@ekiga.net"
target="_blank">cyrenity@ekiga.net</a><br>
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