lakmal, Paul and ALL:
I heard your discussion about best practice of ser sip (maybe 0.9.0 or above) for a long time. And you also said the documents would be posted for everybody to enjoy it. Could you please tell me where to find the documents?
Best Regards Charles
On Wed, 9 Mar 2005 05:45:11 +0100 (CET), lakmal silva ruwan_lakmal@yahoo.fr wrote:
Hi Hans,
This is a great thing. Can't we place this BSS code at berlios? And then may be put a link from iptel.org site under 3rd party software. As Alexey willing to set up a SER community site may be you can place the docs there as well.
By the way are you involved in Yxa and the minisip developments at KTH?
Regards,
Lakmal
--- Hans Eriksson hans@hecc.se wrote:
Guys,
this discussion faded away. Is it still hot but carried out somewhere else?
Anyway, I am committed to put my business support system in the open source. We use it both in projects at Royal Institue of Technology (KTH), Stockholm, as well as commercially (www.xtrafone.com). The BSS handles order, customers, accounts, rating, billing (pre-paid), customer My Pages, etc. The lot!
KTH use SER as its proxies wheras we use the Asterisk in www.xtrafone.com. As long as there are CDRs placed in a MySql tables, the BSS can pick that up and rate, charge.
I have also written a couple of How-To (install ser, mysql etc).
All this I'd like to add to the pot. I am all for to create a complete package for a commercial or non-commercial VoIP operator that includes a proxy, gateway and BSS. And also the docs describing best practices (ser.cfg, logging, NAT traversal, etc). Just add marketing and customers and you roll.
For the BSS I don't really know where to place the code. Beside the ser? sourceforge? It is written in /bin/sh, awk, SQL and php (no, nothing to compile!). Runs on any Linux, MacOSX and FreeBSD system.
Can we get this discussion thread going again and get started putting our stuff into a shared pool where we can get going to change the world (I just could not hold back :-).
/hans
2005-02-21 kl. 14.28 skrev Iqbal Gandham:
Great idea, can I also suggest, and I can help out
if needs (simply
because I am one of them struggling users :-)) is
a debug guide, all
devices seem to have a few quirks to the setup,
and they all seem to
have different setting, eg some support only STUN,
some use the
proxying, others you can dela with at the server
end etc etc, what I
think owuld be useful is a guide to what its
supposed to look like,
i.e the debug log.
I have been through the entire sip syntax , to
figure out where the
messages go/come from, but with the contact
headers, From, and c= I
can see how it can get a little confusing
Iqbal
PS 0.10 works quite nicely
Java Rockx wrote:
Steve, I fully agree - and this is the exact reason that
this cannot be a
single person endeavor. Regards, Paul On Mon, 21 Feb 2005 07:28:02 -0500, Steve Blair blairs@isc.upenn.edu wrote:
Greger V. Teigre wrote:
Paul, I fully support the approach: Make one
reference design with a
complete ser.cfg. This will give us a Getting
Started. We can
later add sections on the more advanced stuff like
redundancy, radius,
etc. Thanks for your review of the components in
such a reference design
(I'll relate to those further below).
I believe there are two hurdles to get on top
of ser: Get a first
working config up and running and then
understanding the concepts
good enough to start tweaking. Many will not have
all the components of
the full reference system you describe, Paul,
so a starting point
with a minimum system is probably needed. I.e. Get
a UA registered
without auth, etc (I see some questions on this too)
I'd like to add a third hurdle, keeping this or
any documentation
up-to-date. One of the biggest issues I've faced is keeping a working, production
supporting, configuration
"correct" across release changes. The situation doesn't get better if there is
alot of out dated
documentation.
In addition to a few core examples I'd suggest a
clearly worded
changelog. The changelog needs to be clearly show what has changed and what is
impacted by the change
on a release by release basis.
$0.02
I thus see the following things that must be
addressed:
- How to read the basic ser.cfg
- The basic ser.cfg, what does it do, what is
the reference design
(is the ser.cfg in cvs appropriate?)
- A description of the reference design with a
"component list"
- The complete ser.cfg
- Conceptual explanations of each logical part
of the ser.cfg
- External systems (Asterisk,
mediaproxy/nathelper), configs, etc
See my inline comments with regards to a
reference design.
> My setup uses SER v0.9 and Asterisk-1.0.2. The
Asterisk server is
> used > __ONLY__ for voicemail because - well lets
face it, Asterisk sucks
> as > a SIP router because it just isn't designed to
be one.
> > So all users are managed by SER and Asterisk
only comes into play
> for > voicemail and for playing recordings such as
"the party you are
> calling has blocked your call" when a call
block is enabled.
We also use 0.9, but does not yet support
voicemail. I think we
should concentrate on 0.9 capabilities and
forget about 0.8.14.
Most people starting up now will probably use 0.9,
at least shortly when
it is released as stable.
Voicemail adds a layer of complexity in terms
of scalability and
redundancy. IMHO we should leave out voicemail
from the reference
design, not because it is something most people
would not want, but
because it introduces an external component and
complexity that is
better added later in the document (like
redundancy). That being
said, I think we should include voicemail and
voiceprompts as part of the
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