[Freeswitch-users] Inbound and then outbound call?

David Ponzone david.ponzone at ipeva.fr
Wed Aug 25 01:33:43 PDT 2010


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
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

Ce message et toutes les pièces jointes sont confidentiels et établis  
à l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autorisée est interdite. Tout message électronique est  
susceptible d'altération. IPeva décline toute responsabilité au  
titre de ce message s'il a été altéré, déformé ou falsifié. Si  
vous n'êtes pas destinataire de ce message, merci de le détruire  
immédiatement et d'avertir l'expéditeur.




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 
> > 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> 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.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100825/c44b83c5/attachment.html 


More information about the FreeSWITCH-users mailing list