[Freeswitch-users] how to pass arbitrary headers from A leg to B leg when bridging

[David Ponzone]

Wouldn't it be simpler to first deal with auth, and to put your transcoding FS after that in your network path ?

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 12/04/2011 à 18:49, Dave Horton a écrit :

> So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now.
> 
> But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity.  First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do.  I basically want to use FS as a simple transcoding server between two endpoints, call them A and B.  Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex.  The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration).  Simple, right?  The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header.  So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg.
> 
> So far, the only way I have found to do that is to hack the code to create a new channel variable.  I've done this, and it works.  However, this leads me to the following questions
> 
> 1) Is there a better way to do this?  If there is no way to do it as a dialplan out of the box, can it be done as a script?
> 
> 2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia.  Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)?
> 
> Dave
> 
> 
> _______________________________________________
> 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/20110426/d9b7e0ca/attachment-0001.html