[Asterisk-Users] Asterisk as SIP Proxy

[ranga]

----- Original Message -----
From: "Olle E. Johansson" <oej at edvina.net>
To: <asterisk-users at lists.digium.com>
Sent: Friday, November 28, 2003 2:57 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy


> ranga wrote:
>
> >
> > I have one linux box running asterisk ( say 192.168.68.15 ) and second
box
> > running partysip (say 192.168.68.6).
>
> > Now this is what I wanted to achieve.
> >     The other sip server ( here partysip) may have many users
registered. It
> > is not possible to make every user's entry into extensions.conf.
Instead,
> > any mechanism where I can replace 192.168.68.6 with a variable that
> > represents the 'To' domain will be a great. In simple, I am looking for
a
> > line in extensions.conf that looks like following.
> > exten => _proxy-.,1,Dial(SIP/${EXTEN:6} @ ${DOMAIN})
> >
> > *variable DOMAIN is my assumption.
> >
> > This is the possible soln that I can think off. This, I guess, needs
little
> > hack into chan_sip.c. Is there any other way that simplifies this task?
> Hello!
>
> Never start with considering a hack in the source code, that only creates
> a mess for yourself when trying to keep updated. It's easier to read
> the documentation :-)
>

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.

> You can create your own variables freely in extensions.conf, see the
sample
> provided in the Asterisk distribution.
>
> Or:
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20readme.variables
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20variables
>

For some reason, I am seeing SIPDOMAIN  blank. Not sure why.

> You can also use ENUM for this. See the tips & tricks page on the same
> server.
>
> The latest CVS version fully supports domain dialling in SIP, so there
> should be no problem with calling SIP/${EXTEN:6}@${DOMAIN}

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.
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.

> Regards,
> /Olle
>
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