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Urgent Technical assistance needed for SIP SoftPhone


 
Urgent Technical assistance needed for SIP SoftPhone Post Reply
24.09.2009 11:55 vijayswami1@online.gmail.com

Hello All,
I am doing very difficult task .So desperately i am looking for technical help.
My Scenario is :
1) I am running one thirdparty NON SIP application on my system.
2) One SIP SoftPhone also is running on my system. Any Softphone take example of X-Lite or Zoiper phone or Ninja Lite etc
3) I have provided one text box and Button for calling in thirdparty application.
4) My requirement is when i will enter the number in text box and after clicking the call button call should go trough that SIP softphone
5) And subsequent response for that INVITE request or call should be indicated to my application
6) Now Whenever SIP phone will get incoming call then i need the indication of that incoming request.
7) And i may receive the call by another button provided in my thord party application. Inshort i want the indication from sip phone to my third party application.

  How this can be achieved? Can we achieve this by using the TeraSIP? If yes,Then can anybody tell me the steps how this can be done.I

Thanks and Regards,
--------------------------
vijay swami
SBSOFT Technology Solutions India Pvt Ltd
vijayswami1@gmail.com
--------------------------
Re: Urgent Technical assistance needed for SIP SoftPhone Post Reply
28.09.2009 23:38 TERASENS Support

Vijay,

does the NON-SIP application support TAPI? Is it an application developed by
you or by another 3rd party?


Best regards,

Matthias Moetje
-------------------------------------
TAPI WIKI: http://www.tapi.info
-------------------------------------
TERASENS GmbH
Augustenstraße 24
80333 Munich, GERMANY
-------------------------------------
e-mail: moetje at terasens dot com
www:   www.terasens.com
-------------------------------------

Re: Urgent Technical assistance needed for SIP SoftPhone Post Reply
29.09.2009 04:57 vijayswami1@online.gmail.com

Hi ,
  Thanks for your reply..Yes Third party application is a TAPI application.I can achive my requirements(as i told earlier) by TERASIP.even i tried also and it works fine.But instead terasip if i want to use some other SIP Soft phones like X-Lite,Ninja Pro,3 CX etc.I need to develop common software for all SIP phones. Please can you suggest me some way to achive this.i am struggling with this problem from last one month.

Regards,
Vijay Swami


--------------------------
vijay swami
SBSOFT Technology Solutions India Pvt Ltd
vijayswami1@gmail.com
--------------------------


>Vijay,
>
>does the NON-SIP application support TAPI? Is it an application developed by
>you or by another 3rd party?
>
>
>Best regards,
>
>Matthias Moetje
>-------------------------------------
>TAPI WIKI: http://www.tapi.info
>-------------------------------------
>TERASENS GmbH
>Augustenstraße 24
>80333 Munich, GERMANY
>-------------------------------------
>e-mail: moetje at terasens dot com
>www:   www.terasens.com
>-------------------------------------
>
>
Re: Urgent Technical assistance needed for SIP SoftPhone Post Reply
29.09.2009 22:33 TERASENS Support

Vijay,

for incoming calls this is rather easy: You can use TERASIP in parallel with
any softphone. It is mandatory for the SIP protocol to allow multiple
simultaneous client registrations (very few providers disallow this,
though).

What you need to do is to set TERASIP or the softphone to a different client
port. E.g. you can change the port on which TERASIP operated from 5060 to
5070. This will allow both the softphone and TERASIP to connect to the
provider.

For outgoing calls: You can create a connection to your softphone and then
perform a SIP refer (call transfer) to the actual destination. To do this
you create a call to the local SIP phone (e.g. call to sip:localhost:5060)
then perform a call transfer (e.g. lineBlindTransfer) to the actual target
(e.g: sip:123456789@sipprovider.com).

Note that this can work also with hardware phones.

PS: A future version of TERASIP will include this "call control" capability
as a configurable option.


Best regards,

Matthias Moetje
-------------------------------------
TAPI WIKI: http://www.tapi.info
-------------------------------------
TERASENS GmbH
Augustenstraße 24
80333 Munich, GERMANY
-------------------------------------
e-mail: moetje at terasens dot com
www:   www.terasens.com
-------------------------------------

Re: Urgent Technical assistance needed for SIP SoftPhone Post Reply
01.10.2009 05:14 vijayswami1@online.gmail.com



Hi,
Thanks a lot for your valuable suggestion.But i have few doubts in your suggestion.

1) For handelling the Outgoing call you have suggested using the call transfer method that is REFER method.I am agrre with that.it can be implemented.I was aware of this method.
2) But for Incoming call only i have doubts:
A) You told for configuring TERASIP and my sip softphone on different ports.Assume i am using SIP Softphone X-Lite on 5060 and TERASIP on 5070.(It means two separate sip softphones are running on my system).
B) Now i will REGISTER both phones with phones with separate public user id e.g. sip:15006@serviceprovider.com.When both phones will register with same user id on server two bindings will be created with same ip address but with different port.
C)So now when some third user will call to user sip:15006@serviceprovider.com then to which phone call INVITE will be sent to TERASIP or X-Lite? (Both have registered with same user id but different port), So server will send the call - INVITE request to that phone which has registered first.If TERASIP has registered first then call will come to terasip or it will go to x-lite.
If call comes to terasip then my application is tapi applicatio. so it will get indication (But in this case x-lite wont be involved at all).
D) Suppose Call came to X-Lite Phone then how it will indicate to terasip or my application. Because Terasip and x-lite are no where connected. they are running on separate port.How they will communicate?

  Please, can you give me some call flow example and exact idea how it can be implented? I am not clear with above points.Please Reply for above doubts.

Thanks and Regards,
--------------------------
vijay swami
SBSOFT Technology Solutions India Pvt Ltd
vijayswami1@gmail.com
--------------------------


>Vijay,
>
>for incoming calls this is rather easy: You can use TERASIP in parallel with
>any softphone. It is mandatory for the SIP protocol to allow multiple
>simultaneous client registrations (very few providers disallow this,
>though).
>
>What you need to do is to set TERASIP or the softphone to a different client
>port. E.g. you can change the port on which TERASIP operated from 5060 to
>5070. This will allow both the softphone and TERASIP to connect to the
>provider.
>
>For outgoing calls: You can create a connection to your softphone and then
>perform a SIP refer (call transfer) to the actual destination. To do this
>you create a call to the local SIP phone (e.g. call to sip:localhost:5060)
>then perform a call transfer (e.g. lineBlindTransfer) to the actual target
>(e.g: sip:123456789@sipprovider.com).
>
>Note that this can work also with hardware phones.
>
>PS: A future version of TERASIP will include this "call control" capability
>as a configurable option.
>
>
>Best regards,
>
>Matthias Moetje
>-------------------------------------
>TAPI WIKI: http://www.tapi.info
>-------------------------------------
>TERASENS GmbH
>Augustenstraße 24
>80333 Munich, GERMANY
>-------------------------------------
>e-mail: moetje at terasens dot com
>www:   www.terasens.com
>-------------------------------------
>
>
Re: Urgent Technical assistance needed for SIP SoftPhone Post Reply
01.10.2009 21:51 TERASENS Support

Vijay,

with the SIP protocol there are different URIs involved:

The AOR URI: e.g. 123456@provider.com
and the contact URIs: e.g.: 123456@12.13.14.15:5060

When both clients will register, there will be 2 server bindings,
i.e. the AOR 123456@provider.com is bound to

123456@12.13.14.15:5060 and
123456@12.13.14.15:5070

SIP calls are made to the AOR URI, the provider's SIP proxy
will then FORK the call to both registered contact URIs
(which means the proxy creates two invite sessions).

This is the way the SIP protocol works. It is MANDATORY for
a SIP proxy to perform this kind of forking.

Some providers disallow multiple registrations, but this is
a violation of the SIP specification!


Best regards,

Matthias Moetje
-------------------------------------
TERASENS GmbH
Augustenstraße 24
80333 Munich, GERMANY
-------------------------------------
e-mail: moetje at terasens dot com
www:   www.terasens.com
-------------------------------------



 
 
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