Inbound and then outbound call?
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). 2. Now, I would like to dial out to a PSTN number so that received call is connected to this new outbound number. How can this be done? Do I use Originate from within my .NET (mod_managed) code? 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? Please help. Thank you.
Malay, in most countries, you are not charged for a call you receive. David Ponzone Direction Technique email: david.ponzone@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 24/08/2010 à 20:58, Malay Thakershi a écrit :
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).
2. Now, I would like to dial out to a PSTN number so that received call is connected to this new outbound number.
How can this be done? Do I use Originate from within my .NET (mod_managed) code?
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?
Please help.
Thank you. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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.
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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@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@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com
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@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@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@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@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@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@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@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@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@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@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@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@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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callwithus? On Wednesday 25 August 2010, 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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user s http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Provide link to your business. On Wed, Aug 25, 2010 at 1:48 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
callwithus?
On Wednesday 25 August 2010, 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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-user s http://www.freeswitch.org
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-- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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@ipeva.fr <mailto:david.ponzone@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@ipeva.fr <mailto:david.ponzone@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@gmail.com <mailto:mthakershi@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@gmail.com <mailto:acosgrov@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@ekiga.net <mailto:cyrenity@ekiga.net> mail: mustafa.pk <http://mustafa.pk>@gmail.com <http://gmail.com> web: cyrenity.wordpress.com <http://cyrenity.wordpress.com> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________
That looks good for incoming DIDs. Any idea about Vonage Small Business Premium Unlimited plan? It does provide SIP option / integration. - Will it let me make simultaneous outbound calls from my FS box? On Wed, Aug 25, 2010 at 2:04 PM, Michael Scheidell < michael.scheidell@secnap.com> wrote:
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@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@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@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@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@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300
*| *SECNAP Network Security Corporation
- Certified SNORT Integrator - 2008-9 Hot Company Award Winner, World Executive Alliance - Five-Star Partner Program 2009, VARBusiness - Best in Email Security,2010: Network Products Guide - King of Spam Filters, SC Magazine 2008
------------------------------
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Hmmm, I suspect you can make 2 calls, but I I would guess no more. The reason is it's an unlimited plan, and if they allow you to send 10 calls at the same time, their business model will be in jeopardy. David Ponzone Direction Technique email: david.ponzone@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 26/08/2010 à 08:48, Malay Thakershi a écrit :
That looks good for incoming DIDs.
Any idea about Vonage Small Business Premium Unlimited plan? It does provide SIP option / integration. - Will it let me make simultaneous outbound calls from my FS box?
On Wed, Aug 25, 2010 at 2:04 PM, Michael Scheidell <michael.scheidell@secnap.com
wrote: 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@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@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300
| SECNAP Network Security Corporation Certified SNORT Integrator 2008-9 Hot Company Award Winner, World Executive Alliance Five-Star Partner Program 2009, VARBusiness Best in Email Security,2010: Network Products Guide King of Spam Filters, SC Magazine 2008
This email has been scanned and certified safe by SpammerTrap®. For Information please see http://www.secnap.com/products/spammertrap/
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participants (6)
-
Anthony Cosgrove
-
David Ponzone
-
Ghulam Mustafa
-
Malay Thakershi
-
Michael Scheidell
-
Sergey Okhapkin