asterisk users - Mar 2006 - Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?

Hello,

I use both ser/asterisk .

In fact i wish asterisk to forward all the sip
requests  which are not handled by domain=domain.tld
in sip.conf 

Here is a diagram:

The sip agents use the Sip proxy as an outbound sip
proxy and domain=domain.tld  .

When the sip agents dial sip:user@otherdomain.tld so
the request is sent to sip proxy and so to Asterisk.
I wish Asterisk to Look up the domain's part of the
uri.
if the domain is not equal to domain.tld the request
is sent back to the sip proxy.

Is it possible ?

   ASTERISK 
    |    | 
    |    |
 ==SIP proxy===sip agents

Thanks for help

Harry
  
--- Wilmar Campos <wilmar.campos@gmail.com> a ?crit :
> Yes:
> 
> 
> ;
> ; Provider or Remote PEER
> ;
> register => 800:345698:12345678@sip_provider/800
> 
> [sip_provider]
> type=peer
> context=default
> ;secret=345698
> fromuser=800
> host=sip.provider.com
> ;language=es
> dtmfmode=rfc2833
> disallow=all
> allow=g729
> canreinvite=no
> I hope this is what you are looking for.
> 
> Regards,
> 
> Wilmar
> 
> 
> On 3/7/06, hgaillac-sip@yahoo.fr
> <hgaillac-sip@yahoo.fr> wrote:
> >
> > Hello,
> >
> > I posted this question to
> > asterisk-users@lists.digium.com
> > without reply.
> >
> > Is there a way to define an outbound proxy in
> sip.conf
> > ?
> > I wish to forward the INVITE requests to an
> outbound
> > proxy when Asterisk (1.2.x) doesn't handle the
> domain.
> >
> >
> > Regards
> > Harry
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
>
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> 







	

	
		
___________________________________________________________________________ 
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs
exceptionnels pour appeler la France et l'international.
T?l?chargez sur http://fr.messenger.yahoo.com

On 3/7/06, hgaillac-sip@yahoo.fr <hgaillac-sip@yahoo.fr>
wrote:
>
>
> When the sip agents dial sip:user@otherdomain.tld so
> the request is sent to sip proxy and so to Asterisk.
> I wish Asterisk to Look up the domain's part of the
> uri.
> if the domain is not equal to domain.tld the request
> is sent back to the sip proxy.
>
> Is it possible ?
>
>
You can check the ${SIPDOMAIN} variable in the dialplan on an incoming call
and then do whatever you want in the dialplan based on that (i.e. exten =>
123,1,GotoIf($["${SIPDOMAIN}" :
"otherdomain.tld"]?context|${EXTEN}|1:2) or
something like that and then have a Dial statement directed at the SER box
at context|${EXTEN}|1).  Keep in mind though that this is a completely new
call.  Asterisk is not a SIP proxy so you can't just re-route the SIP packet
back to SER.  It will have a new call-id, etc.

Terry
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Hey everyone,
    I am working with Verizon to terminate SIP calls for me.  

MY QUESTION:  I notice the diagram shows only one asterisk server (softswitch). 
Since asterisk is not very good at clustering (appearing as one entity), I would
have to have multiple servers (softswitches).  You mentioned "You will give
us the IP addresses of your softswitch(es) and that will allow communication
between your network and ours."  If I gave you 3 IP addresses (to three
asterisk softswitches), how would you hand off the calls unless there was some
policy defining which calls go to which IPs.

THEIR RESPONSE: I only represented one softswitch, because many of my previous
customers only have one.  Since you will have multiple softswitches, do you plan
on implementing a session border controller?  This would allow us to point the
traffic to the SBC and thereby eliminating the worry of trying to separate the
traffic.  Are all the Asterisks going to be in the same location?

    Soooo, what does she mean by a session border controller?  As I understand,
* is the SBC right?  If not, how would an SBC fit into the picture with
Asterisk?


- Gabe

P.S. And yes, I have thought about SER as a solution to this and have not yet
given it a try.

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It sounds like they referring to a load balancer that would provide one live
external IP and handle sessions for multiple internal servers.


_____________________
Mobilcom
http://www.mobilcom.net


  ----- Original Message ----- 
  From: Gabriel Afana 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Friday, March 10, 2006 2:11 AM
  Subject: [Asterisk-Users] Some ignorance here,what exactly is a Session
Control Border? (Verizonis asking me about this)


  Hey everyone,
      I am working with Verizon to terminate SIP calls for me.  

  MY QUESTION:  I notice the diagram shows only one asterisk server
(softswitch).  Since asterisk is not very good at clustering (appearing as one
entity), I would have to have multiple servers (softswitches).  You mentioned
"You will give us the IP addresses of your softswitch(es) and that will
allow communication between your network and ours."  If I gave you 3 IP
addresses (to three asterisk softswitches), how would you hand off the calls
unless there was some policy defining which calls go to which IPs.

  THEIR RESPONSE: I only represented one softswitch, because many of my previous
customers only have one.  Since you will have multiple softswitches, do you plan
on implementing a session border controller?  This would allow us to point the
traffic to the SBC and thereby eliminating the worry of trying to separate the
traffic.  Are all the Asterisks going to be in the same location?

      Soooo, what does she mean by a session border controller?  As I
understand, * is the SBC right?  If not, how would an SBC fit into the picture
with Asterisk?


  - Gabe

  P.S. And yes, I have thought about SER as a solution to this and have not yet
given it a try.
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