Hi Jiri,
Sure I can do what you said. I mentioned that in my original message too.
when sathya(a)foo.bar calls sathya(a)bar.foo , I need bar.foo proxy to think
that the call originator is jiri(a)bar.foo. Just like terminating into PSTN,
where PSTN is always thinks that the call is originated from PSTN.
I proxy needs to be a stateful for this kind of an application as RTP path
may have to go through the proxy.
Cheers
Sathya
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Tuesday, October 21, 2003 1:20 PM
To: Sathya Weerasooriya; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Trunking between two sip networks
What's the problem here? You need to have IP routing in place and
that's it.
sathya(a)foo.bar can then call sathya(a)bar.foo at will very much like with
email. If devices are capable only of numbers, you need to set up
a dialing
plan, that's it.
-jiri
At 07:31 PM 10/19/2003, Sathya Weerasooriya wrote:
Hi Folks,
I am not a SER user yet, but hopefully soon.
I have a need to interconnect two SIP networks just like we use a SIP to
PSTN gateway (to connect SIP network and PSTN).
Let me explain this senario in simple terms.
Lets say I have 1 port FXO g/w. Then all my clienst in the ip network can
use only one PSTN line to get to PSTN. And PSTN always thinks
that the calls
are originated from the directory number of this
PSTN line.
Similarly lets say I have 1 phone number, 1234(a)iptel.org ( one
account from
iptel). And I have my own network of Phones in a
private SIP
network. I want
my clients to be able to talk to IPtel network
just as the above senario,
where IPtel would think the call is originated from 1234(a)iptel.org. True
only one call at a time, just like PSTN senario.
How could this be done ?
BTW : This is not to make my phones in private network to have
the ability
to talk to IPtel phones. I can do that with my
next-hop proxy setting and
ip-dialplan.
Cheers
Sathya
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Jiri Kuthan
http://iptel.org/~jiri/