21 May
2005
21 May
'05
10:17 p.m.
Hi
This sounds a reverse setup to what I have, or am trying to do, so am
interested as to the thought process behind this.
Your users are registered to asterisk, several boxes, I presume u r
pulling sip.conf from a central DB, that way they can hit any box, and
now to route out to pstn you wish to have pstn providers in various
countries. So asterisk will decide that its a non-ip call, on non-local,
i,e where ur asterisk cannot terminate, and then send out to remote
gateway, you could just add all this as a dial channel in each asterisk
box, or use asterisk lcr module (I havent used this) to do the routing
for you.
Yes, I could do this and it's possible that I'll have to do it this way, but
I'd prefer to
have a central box(es) that roughly speaking interfaces outside world.
As greg mentioned you could do this with ser and lcr
adding your gateways
as and where you want, now the problem you may hit here is that there
are a handful of gateways which take IP auth, and of the few who do, are
H323, which means that passing a call out to ser may not work, since it
will not recognise the h323 (if u decide to sip-->h323 convert) that is
coming from asterisk.
I understand it. I'm talking about SIP termination.
As I mentioned this seems to be a reverse scenario to
my setup, where I
am regsitering all the phones with ser, which I think is better than
doing it at asterisk, simply because it seems to handle it better (but
it dont do H323), and then (and this is in beta) what I am doing is do
detect if the user is a corporate, and allow him pbx functionality i.e
extensions settings etc etc.
[PSTN connectivity is via SER, and done via IP auth, however this does
mean inbound from pstn goes ser ---> asterisk--> ser---pstn if I need to
forward call back out to a pstn number..but I cant work out a better
way.]
Now what this allows is to is to bypass asterisk completely for the main
residential users, and even ignore the media stream completely, for
those users who just wish to do cheap calling.
Well, my scenario doesn't exclude it. If I need just allow cheap calling for
residential users, I'll set up a ser box and make it talk to everything else.
Asterisk gets pulled in when I know a user has paid me
more :-). The
problem with some gateways only allowing H323 I am looking to solve by
using lcr to route to my asterisk box, and from there doing sip->h323
and then send out to the gateway, i know this is not the best solution
since it will add overhead, but I dont have a better one as yet.
Just remember, with sip<->h323 translation in asterisk, media will always go
through asterisk. I recently saw another project at http://yate.null.ro that claims
to be able to do it while sending media stream directly between endpoints.
Not sure if that helps, its my saturday morning
rambling before the FA
Cup final....come on Utd (for those not from the UK...its like the
superbowl, and for those not in the US...I give up :-))
Would like to know why register to asterisk even though using pbx, you
could route those users through...unless I am missing something
For many reasons, including historical ones. Also 'cause it solves many problems
almost out of the box that I'm sure you and others trying integrate ser and
asterisk are experiencing, like voicemail indication, nat traversal, per context
music on hold, very flexible database driven dialplans etc. The system was mostly
designed for business customers as hosted PBX offering and as I said if we decide
to go into residential market for just cheap calling I think it can be done with
reasonable
efforts.
Iqbal
On 5/20/2005, "Michael Ulitskiy" <mdu113(a)acedsl.com> wrote:
--
See you later,
Michael
-------------------------------------------------------
--
See you later,
Michael
Can it do what I described?
Documentation isn't very good on that web site, but according to what I saw
I doubt it.
Also what does it mean SER based PBX? SER+voicemail?
Thanks,
Michael
On Friday 20 May 2005 07:20 pm, you wrote:
we have an all in one ser solution that might be usefull for you
http://www.wifi.com.ar/english/voip.html
regards,
____________________________________________________________________________
Jaime Garcia Ghirelli http://www.brujula.net
jaime(a)fonosip.com http://www.fonosip.com
http://www.wifi.com.ar
---------- Forwarded message ----------
Date: Fri, 20 May 2005 15:49:35 -0400
From: Michael Ulitskiy <mdu113(a)acedsl.com>
To: serusers(a)lists.iptel.org
Subject: [Serusers] Advice needed
Hello,
I'd like ask for advice on what is in your opinion the best solution
in the following scenario.
I have a bunch of sip servers (asterisk boxes as my users need pbx
functionality) that can make sip call to each other and my PSTN
gateway. Now I want to purchase PSTN terminitaion in several
different markets (and probably more in the future). All those
terminations will require authentication.
I want all my boxes when they see non-local call to send it to a
central routing server that would determine where this call should
be sent and authenticate to the appropriate provider so that I don't
have to configure all credentials on all asterisk boxes. Also I want
it not to deal with the media at all. All media streams should go directly
from asterisk box to the PSTN termination provider.
So basically it should be central SIP router that is able to authenticate
calls if neccessary.
I thought I could do it with SER and its UAC module, but it appears
UAC module doesn't work and probably won't work (see my previous
post in this list about UAC backport to 0.9.0).
Also I don't want to use asterisk in this place as asterisk always wants to
stay in media path and I'd really like to avoid of getting into hassle with
re-invites.
So the question is what are my options and what you would advice
as a solution. Are there any software out there that can do it (preferably
open-source, of course) or what else you could suggest to do to get
desired results.
Thanks a lot,
--
See you later,
Michael
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--
See you later,
Michael
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