[Freeswitch-users] Inbound and then outbound call?
Michael Scheidell
michael.scheidell at secnap.com
Wed Aug 25 12:04:02 PDT 2010
don't know vitelity,
but voip.ms. supports user/password or ip based auth. has built in
Vmail, redirect on network outage.
us did's are .99/c per month, and calls 1.1c/ month. (20? trunks per did?)
On 8/25/10 2:28 PM, Malay Thakershi wrote:
> Can someone suggest me wholesale and cheaper provider than Vitelity
> but offering similar service?
>
> In/Out bound both USA.
>
> Thank you.
>
> On Wed, Aug 25, 2010 at 3:33 AM, David Ponzone <david.ponzone at ipeva.fr
> <mailto:david.ponzone at ipeva.fr>> wrote:
>
> AFAIK, most (all ?) providers wont accept the transfer.
> They are not allowed to change the rate of the call from the
> caller perspective after the call was initiated.
> Imagine you transfer the call to an Iridium number...
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service Client IPeva
> tel: 0811 46 26 26
> www.ipeva.fr - www.ipeva-studio.com
>
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>
> Le 25/08/2010 à 07:44, Ghulam Mustafa a écrit :
>
>> Hi,
>>
>> 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
>>
>> :/
>>
>> -mustafa
>>
>> On Wed, Aug 25, 2010 at 10:33 AM, Malay Thakershi
>> <mthakershi at gmail.com <mailto:mthakershi at gmail.com>> wrote:
>>
>> I am currently looking at Vitelity. They have 1.44 cents per
>> minute. They do charge for incoming/outgoing both.
>>
>> I am not literally dialing out to a phone. I want to dial a
>> mobile number or any other US number for that matter.
>>
>> Here is what I intent to do:
>> 1. A US phone dials to Vitelity number -- comes to my FS box
>> 2. I validate few things
>> 3. Dial another US number and connect received call to that one
>>
>> So as you said I will be charged 2 * 1.44 (since I can't
>> terminate the arrived call after validation).
>>
>> Is there any other way to the sequence I have specified above?
>>
>> Are there providers similar to Vitelity but cheaper (with
>> relatively same features)?
>>
>> Thank you.
>>
>> On Tue, Aug 24, 2010 at 6:06 PM, Anthony Cosgrove
>> <acosgrov at gmail.com <mailto:acosgrov at gmail.com>> wrote:
>>
>> On Tue, 2010-08-24 at 13:58 -0500, Malay Thakershi wrote:
>> > Hello,
>> >
>> >
>> > It would be a great help if someone can guide me.
>> >
>> >
>> > 1. I would like to first receive a call, perform
>> certain validations.
>> > (Able to do this via mod_managed application that
>> handles call from
>> > dialplan).
>> >
>>
>> That should not be a problem, I don't know your
>> requirements so can't
>> provide a full answer.
>>
>> >
>> > 2. Now, I would like to dial out to a PSTN number so
>> that received
>> > call is connected to this new outbound number.
>> >
>>
>> This can be done and is called hairpinning.
>>
>> >
>> > How can this be done? Do I use Originate from within my
>> .NET
>> > (mod_managed) code?
>> >
>>
>> Yes, you would be bridging the two legs like any normal
>> call. Instead of
>> going to an endpoint you're going back out over the PSTN.
>>
>> >
>> > Do I get charged for both incoming and outbound call
>> until the entire
>> > session ends? Is there a way to receive call, validate
>> and then sort
>> > of transfer and then terminate the received call so I
>> do not get
>> > charged for both?
>>
>> That would depend on your provider but most likely yes.
>> As for
>> terminating one end after validation that is not going to
>> happen. A leg
>> terminates to you on an agreed fee schedule. No carrier
>> that I know of
>> supports that. Now if you kept everything SIP... you
>> could do a transfer
>> after the validation.
>>
>>
>>
>> Anthony C.
>>
>>
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>>
>>
>> --
>> Ghulam Mustafa
>> cell: +92 333.611.7681
>> sip: cyrenity at ekiga.net <mailto:cyrenity at ekiga.net>
>> mail: mustafa.pk <http://mustafa.pk>@gmail.com <http://gmail.com>
>> web: cyrenity.wordpress.com <http://cyrenity.wordpress.com>
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--
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