Re: [OpenSIPStack] Help using OpenSipStack / Problems answering a SIP call

[Christian W. <cwa...@gm...>]

Hi Ilian,


I got it, the problem was that I sent the AnswerCall before the stack has
passed all OnIncomingConnection events (I have sent the answercall in 
OnIncomingConnection which seams to be wrong)...

Thanks for your help !


Kind regards 

Christian

-----Ursprüngliche Nachricht-----
Von: ope...@li...
[mailto:ope...@li...] Im Auftrag von
Ilian Jeri C. Pinzon
Gesendet: Montag, 12. Mai 2008 11:19
An: ope...@li...
Betreff: Re: [OpenSIPStack] Help using OpenSipStack / Problems answering a
SIP call

Hi,

Since you won't use the PC sound system, the modifications for what you 
need are not trivial.

First, you need to re-enable the following lines in SoftPhone.cxx:

AddRouteEntry("pc:.*  = sip:<da>");
AddRouteEntry("sip:.*  = pc:<da>");

These sets the linkages of  the endpoints. In this case, 
SIP-to-PCsoundsystem and PCsoundsystem-to-SIP.

Then you will have to override OpalEndPoint and OpalConnection. You can 
use OpalPCSSEndpoint and OpalPCSSConnection as reference. But instead of 
using OpalAudioStream, your classes will need to use a specialized 
OpalMediaStream that can read from your source and write any incoming 
data.  OpalFileMediaStream will probably suffice if you play and record 
from files.  To get a grasp of how the process works, try setting up 
break points in OpalPCSSEndpoint, OpalPCSSConnection , and OpalMediaStream.

Christian Wallukat wrote:
> Hi all,
>
>
>
> I need urgent Help, since I didnt get it ...
> I attached the SoftPhone files which I have 
> Modified. For my project I only need to get the 
> Call and access to the media stream (recording
> And playing).
>
>
>   
>


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