[Asterisk-Users] Asterisk as SIP Proxy

[Olle E. Johansson]

ranga wrote:

> I agree with you. But the issue is, how could I fix the domain name
> variable? This should not be static. The target domain changes as per the
> choice of the user that is connected through softphone. For example, you are
> connected to edvina.net. Now I want to call you from my softphone. I have a
> SIP account ranga at myprovider.com. This demands me to add your domain in the
> configuration of  myprovider.com. This server might have a many users and
> everybody needs a service extended to the other users connected to other
> domains that are running non-asterisk servers. So, everytime a new domain is
> requested for dial, the asterisk admin need to add that domain explicitly.
> This makes his job tedius.
> 
> So, I thought setting DOMAIN variable to the target domain in chan_sip.c
> would help. Not sure of complications.
To call me you don't have to define "edvina.net" in the asterisk server.
Dial(SIP/oej at edvina.net)
works all right.
The problem to fix is when a client, like x-lite, dials "sip:oej at edvina.net".
Asterisk treats this incoming SIP call as a call to "oej"
check the SIPDOMAIN variable to get the "edvina.net" part and put
them back together again.
DIAL(SIP/${EXTEN}@${SIPDOMAIN}) should fix it. Just watch out to check
if SIPDOMAIN is the realm of your Asterisk server before dialing out.

> 
> I checked it out on 26th of Nov. Any updates in this couple of days towards
> this?
> 
> 
>>If I misunderstood you, please explain a bit more so we can help you.
> 
> 
> Its like this: I saw domain dialing in SIP working. When we dial SIP ID from
> softphone, asterisk considers the part before '@' as extension. So, we will
> need to specifically mention the domain in the call to Dial application.
> This is what I wanted to avoid. I would like to pick it from the INVITE
> request.
That is how it works today.

> In this case, I can have a standard way of representing the other domain
> IDs. For example if I want to call you through my asterisk box, I wil call
> you as
> <sip:proxy-oej at edvina.net>. This way I will not need to mention your domain
> name explicitly in the extensions.conf.
I dial domains from X-lite connected to my ASterisk and it works. If I just
enter "10122", X-lite adds the default SIP realm and the server recognizes
this as a local extension by checking SIPDOMAIN. If it's not the local
SIP realm (like "sip:21343 at fwd.pulver.com", I add the SIPDOMAIN (as above)
and it dials out by checking  DNS SRV records.

I'll add an example to the Wiki later.

/Olle