[Asterisk-Users] Registrar only setup

[Tomas Florian]

No I haven't tried it - but looks like exactly what I'm missing. 

 

Thanks Ariel !

 

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ariel Batista
Sent: Tuesday, August 30, 2005 7:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Registrar only setup

 

have you tried in the sip.conf for the devices

 

canreinvite=yes

----- Original Message ----- 

From: Tomas Florian <mailto:tflorian at telus.net>  

To: asterisk-users at 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


  _____  


_______________________________________________
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050830/5a374f0c/attachment.htm