Hi all my network topology: endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101) endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103 my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101 can FS do it Thank you __________________________________________________ Bạn Có Sử Dụng Yahoo! Không? Mệt mỏi vì thư rác? Yahoo! Thư có chương trình bảo vệ chống thư rác hữu hiệu nhất trên mạng http://vn.mail.yahoo.com
what is a sip server? Yes, you can do pretty much anything like this. On Apr 19, 2010, at 12:08 AM, false wrote:
Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
Thank you
__________________________________________________ Bạn Có Sử Dụng Yahoo! Không? Mệt mỏi vì thư rác? Yahoo! Thư có chương trình bảo vệ chống thư rác hữu hiệu nhất trên mạng http://vn.mail.yahoo.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
Hi Michael Jerris my sip server is cisco sip server could you guide me some clue or show the same config for the case Thank you Ha` ________________________________ Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR what is a sip server? Yes, you can do pretty much anything like this. On Apr 19, 2010, at 12:08 AM, false wrote: Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
Thank you
__________________________________________________ Bạn Có Sử Dụng Yahoo! Không? Mệt mỏi vì thư rác? Yahoo! Thư có chương trình bảo vệ chống thư rác hữu hiệu nhất trên mạng http://vn.mail.yahoo.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
I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do? Mike On Apr 25, 2010, at 10:44 PM, false wrote:
Hi Michael Jerris
my sip server is cisco sip server could you guide me some clue or show the same config for the case
Thank you Ha` Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
what is a sip server? Yes, you can do pretty much anything like this.
On Apr 19, 2010, at 12:08 AM, false wrote:
Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
Hi Mike let me make it clear :) ip phone1 (ext 1000) registers to SIP server ip phone2 (ext 1002) registers to SIP server FS (ext 1003) register to SIP server the media proxy and sip server are in the same server the signaling flow : ext1000 ---> INVITE ---> SIP server ---> INVITE---> ext 1002 the media flow: ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002 so the signaling for FS when i call 1003 step1: ext1000--->INVITE--->Sip server---->INVITE---->FS step2: ext1000 press 1002, FS will transfer call to SIP server signaling flow for step 2: 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold 2. FS --->INVITE---> SIP server----> INVITE--->ext1002 when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 3. ext1000 connected to ext1002 media flow: step1:ext1000----RTP---media proxy----RTP---FS step2: 1. FS -----no RTP---media proxy----no RTp----ext1000 2. FS----RTP----media proxy---ext1002 3 ext1000----media proxy----ext1002 Thank you ________________________________ Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 11:23:25, Thứ Hai, 26 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do? Mike On Apr 25, 2010, at 10:44 PM, false wrote: Hi Michael Jerris
my sip server is cisco sip server could you guide me some clue or show the same config for the case
Thank you Ha`
________________________________ Từ: Michael Jerris <mike@jerris.com>
Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
what is a sip server? Yes, you can do pretty much anything like this.
On Apr 19, 2010, at 12:08 AM, false wrote:
Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
Well.. I still don't get what a "sip server" is. Lets just call it a figgledingerbort, that term seems just as useful and defined. You could use the deflect app to do what you seem to be suggesting. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri. Mike On Apr 26, 2010, at 11:40 PM, false wrote:
Hi Mike
let me make it clear :)
ip phone1 (ext 1000) registers to SIP server ip phone2 (ext 1002) registers to SIP server FS (ext 1003) register to SIP server
the media proxy and sip server are in the same server
the signaling flow : ext1000 ---> INVITE ---> SIP server ---> INVITE---> ext 1002
the media flow: ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002
so the signaling for FS when i call 1003 step1: ext1000--->INVITE--->Sip server---->INVITE---->FS step2: ext1000 press 1002, FS will transfer call to SIP server signaling flow for step 2: 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold 2. FS --->INVITE---> SIP server----> INVITE--->ext1002 when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 3. ext1000 connected to ext1002
media flow: step1: ext1000----RTP---media proxy----RTP---FS step2: 1. FS -----no RTP---media proxy----no RTp----ext1000 2. FS----RTP----media proxy---ext1002 3 ext1000----media proxy----ext1002
Thank you
Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 11:23:25, Thứ Hai, 26 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do?
Mike
On Apr 25, 2010, at 10:44 PM, false wrote:
Hi Michael Jerris
my sip server is cisco sip server could you guide me some clue or show the same config for the case
Thank you Ha` Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
what is a sip server? Yes, you can do pretty much anything like this.
On Apr 19, 2010, at 12:08 AM, false wrote:
Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
Are there fries availabkle with a figgledingerbort ? Am 27.04.2010 um 06:02 schrieb Michael Jerris:
Well.. I still don't get what a "sip server" is. Lets just call it a figgledingerbort, that term seems just as useful and defined. You could use the deflect app to do what you seem to be suggesting.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri.
Mike
On Apr 26, 2010, at 11:40 PM, false wrote:
Hi Mike
let me make it clear :)
ip phone1 (ext 1000) registers to SIP server ip phone2 (ext 1002) registers to SIP server FS (ext 1003) register to SIP server
the media proxy and sip server are in the same server
the signaling flow : ext1000 ---> INVITE ---> SIP server ---> INVITE---> ext 1002
the media flow: ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002
so the signaling for FS when i call 1003 step1: ext1000--->INVITE--->Sip server---->INVITE---->FS step2: ext1000 press 1002, FS will transfer call to SIP server signaling flow for step 2: 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold 2. FS --->INVITE---> SIP server----> INVITE--->ext1002 when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 3. ext1000 connected to ext1002
media flow: step1: ext1000----RTP---media proxy----RTP---FS step2: 1. FS -----no RTP---media proxy----no RTp----ext1000 2. FS----RTP----media proxy---ext1002 3 ext1000----media proxy----ext1002
Thank you
Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 11:23:25, Thứ Hai, 26 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do?
Mike
On Apr 25, 2010, at 10:44 PM, false wrote:
Hi Michael Jerris
my sip server is cisco sip server could you guide me some clue or show the same config for the case
Thank you Ha` Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR
what is a sip server? Yes, you can do pretty much anything like this.
On Apr 19, 2010, at 12:08 AM, false wrote:
Hi all
my network topology:
endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101)
endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103
my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101
can FS do it
_______________________________________________ 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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hi folks, If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it. Jan _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969
Hi, I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing, HTH Phil If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me
know how you did it.
Jan
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Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java. As a matter of interest how do you communicate with C# ? --- One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use. Jan Date: Tue, 27 Apr 2010 08:13:59 -0400 From: pjintheusa@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL Hi, I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing, HTH Phil If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it. Jan Hotmail: Trusted email with powerful SPAM protection. Sign up now. _______________________________________________ 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 _________________________________________________________________ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969
You can use mod_managed and run managed code on top of an active call. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2010-04-27, at 10:14 AM, Jan Berger wrote:
Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java.
As a matter of interest how do you communicate with C# ?
---
One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use.
Jan
Date: Tue, 27 Apr 2010 08:13:59 -0400 From: pjintheusa@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
Hi,
I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing,
HTH
Phil
If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.
Jan
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Thanks, I will look into this one. From: mrene_lists@avgs.ca Date: Tue, 27 Apr 2010 10:17:38 -0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL You can use mod_managed and run managed code on top of an active call. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca On 2010-04-27, at 10:14 AM, Jan Berger wrote: Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java. As a matter of interest how do you communicate with C# ? --- One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use. Jan Date: Tue, 27 Apr 2010 08:13:59 -0400 From: pjintheusa@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL Hi, I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing, HTH Phil If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it. Jan Hotmail: Trusted email with powerful SPAM protection. Sign up now. _______________________________________________ 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 Hotmail: Free, trusted and rich email service. Get it now. _______________________________________________ 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 _________________________________________________________________ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969
You might also consider the event socket (just search esl in the wiki) I am toying with this as it shields us from any future changes that break mod_managed. I really like mod_managed though and for me it works great so far. On Tue, Apr 27, 2010 at 11:04 AM, Jan Berger <janvb@live.com> wrote:
Thanks, I will look into this one.
------------------------------ From: mrene_lists@avgs.ca Date: Tue, 27 Apr 2010 10:17:38 -0400
To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
You can use mod_managed and run managed code on top of an active call.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene@avgs.ca
On 2010-04-27, at 10:14 AM, Jan Berger wrote:
Thanks - yes I am aware that you can access wsdl specs from C#, C, C++ or Java.
As a matter of interest how do you communicate with C# ?
---
One of the alternatives I am toying with is to use gSOAP to expose FreeSWITCH API as a WS. It's a bit of work, but it would solve a lot of the issues I see with using FS in real projects. Have not looked into this, and need to see if there is any standards I can use.
Jan
------------------------------ Date: Tue, 27 Apr 2010 08:13:59 -0400 From: pjintheusa@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS connecting to WSDL/SOAP/BPEL
Hi,
I have done this using scripting. managed/c# in my case. Basically all inbound calls are passed to a script - get-session-data, This script calls a ws and populates session variables with the result. I then pass the call back to the dialplan for further processing,
HTH
Phil
If anyone have connected FreeSWITCH to WSDL/BPEL based WebService - let me know how you did it.
Jan
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I want a red figgledingerbort :) From: mike@jerris.com Date: Tue, 27 Apr 2010 00:02:52 -0400 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] need help on IVR Well.. I still don't get what a "sip server" is. Lets just call it a figgledingerbort, that term seems just as useful and defined. You could use the deflect app to do what you seem to be suggesting. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect This will send the invite to your figgledingerbort or whatever sip proxy or UA you want to send it to as specified in the uri. Mike On Apr 26, 2010, at 11:40 PM, false wrote: Hi Mike let me make it clear :) ip phone1 (ext 1000) registers to SIP server ip phone2 (ext 1002) registers to SIP server FS (ext 1003) register to SIP server the media proxy and sip server are in the same server the signaling flow : ext1000 ---> INVITE ---> SIP server ---> INVITE---> ext 1002 the media flow: ext1000 ----RTP ------>media proxy(sip server)----RTP----->ext1002 so the signaling for FS when i call 1003 step1: ext1000--->INVITE--->Sip server---->INVITE---->FS step2: ext1000 press 1002, FS will transfer call to SIP server signaling flow for step 2: 1. FS --->INVITE ---> SIP server---->INVITE--->ext1000 - to put ext1000 on hold 2. FS --->INVITE---> SIP server----> INVITE--->ext1002 when FS gets 200ok from ext1002 then FS --->refer--->Sip server--->ext1000 - FS does the call transfer 3. ext1000 connected to ext1002 media flow: step1: ext1000----RTP---media proxy----RTP---FS step2: 1. FS -----no RTP---media proxy----no RTp----ext1000 2. FS----RTP----media proxy---ext1002 3 ext1000----media proxy----ext1002 Thank you Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 11:23:25, Thứ Hai, 26 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR I meant this literally. People always use the term "sip server" and I never understand what they are trying to say. What exactly is a sip server? I think you have your terms all completely confused in your explanation below or I am just being really dense. This cisco server is a sip server and a media proxy? Can you try to be very clear on what exactly these things are and what you are trying to do? Mike On Apr 25, 2010, at 10:44 PM, false wrote: Hi Michael Jerris my sip server is cisco sip server could you guide me some clue or show the same config for the case Thank you Ha` Từ: Michael Jerris <mike@jerris.com> Đến: freeswitch-users@lists.freeswitch.org Gửi ngày: 1:08:26, Chủ nhật, 25 tháng 4 2010 Chủ đề: Re: [Freeswitch-users] need help on IVR what is a sip server? Yes, you can do pretty much anything like this. On Apr 19, 2010, at 12:08 AM, false wrote: Hi all my network topology: endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101) endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103 my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101 can FS do it _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969
participants (6)
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false
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Jan Berger
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Mathieu Rene
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Michael Jerris
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Michal Bielicki
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Phillip Jones