asterisk users - Aug 2005 - Registrar only setup

Hello,

 

I'm having trouble figuring out how to setup Asterisk so that it's only
a
registrar - not passing any RTP data during phone calls.

So far I got this far:

 

Asterisk server holds registration information for phones

Phones register with Asterisk giving it their ip+port where they can be
currently contacted

NAT doesn't seem to be a problem because STUN seems to take care of it
nicely for me.

 

The hard part that I don't understand is this:

 

Phones can call each other BUT all the RTP traffic is passed through
Asterisk . I don't want this, I need that the phones call each other
directly based on the registration info stored in Asterisk.  I'm having hard
time wrapping my head around this - I think I'm missing some key part - but
the way I understand Asterisk is that it listens for requests on the SIP
channel, when it gets a request it handles it appropriately using it's dial
plan.  But in the dial plan the only thing that makes sense to use is
"dial"
and once I do that all the RTP is sent through asterisk (in-out) to the
other phone. right?

 

Or maybe the problem is on the phone setup?  I tried to make sure that I'm
not specifying any outbound proxy but I do have to specify "proxy"
otherwise
it will not know where to register . right?  

 

Or maybe I'm all messed up 8-P . I thought I understood asterisk at least a
*bit* until I came across this :-)

 

Thanks for any clarification,

Tomas

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have you tried in the sip.conf for the devices

canreinvite=yes
  ----- Original Message ----- 
  From: Tomas Florian 
  To: asterisk-users@lists.digium.com 
  Sent: Tuesday, August 30, 2005 8:48 PM
  Subject: [Asterisk-Users] Registrar only setup


  Hello,

   

  I'm having trouble figuring out how to setup Asterisk so that it's
only a registrar - not passing any RTP data during phone calls.

  So far I got this far:

   

  Asterisk server holds registration information for phones

  Phones register with Asterisk giving it their ip+port where they can be
currently contacted

  NAT doesn't seem to be a problem because STUN seems to take care of it
nicely for me.

   

  The hard part that I don't understand is this:

   

  Phones can call each other BUT all the RTP traffic is passed through Asterisk
. I don't want this, I need that the phones call each other directly based
on the registration info stored in Asterisk.  I'm having hard time wrapping
my head around this - I think I'm missing some key part - but the way I
understand Asterisk is that it listens for requests on the SIP channel, when it
gets a request it handles it appropriately using it's dial plan.  But in the
dial plan the only thing that makes sense to use is "dial" and once I
do that all the RTP is sent through asterisk (in-out) to the other phone. right?

   

  Or maybe the problem is on the phone setup?  I tried to make sure that I'm
not specifying any outbound proxy but I do have to specify "proxy"
otherwise it will not know where to register . right?

   

  Or maybe I'm all messed up 8-P . I thought I understood asterisk at least
a *bit* until I came across this :-)

   

  Thanks for any clarification,

  Tomas



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