Re: [Freeswitch-users] how to pass arbitrary headers from A leg to B leg when bridging
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
maybe opensips would be more adapted for your needs ----- Original Message ----- From: "Dave Horton" <dave@dchorton.com> To: <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, April 12, 2011 12:49 PM Subject: Re: [Freeswitch-users] how to pass arbitrary headers from A leg toB leg when bridging
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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
-
Dave Horton
-
David Ponzone
-
Madovsky