Does the message "Noisy Feedback tells" mean that the server is
configured to include debugging information in its communications?
Kind of like "verbose" mode?
Hi Jan,
This sounds good. I have another question: How do you deal with dependencies on external libraries. Do you store the libraries in CVS? For the OSP module, the toolkit library is required, and I am wondering what is the best way of providing it to the users.
Also, does IPTell or anyone else distribute pre-compiled versions of SER? I am trying to explore the possiblities of distributing a OSP enabled version of SER for those who dont want to go through the trouble of compiling the OSP library.
Vikrant
Maxim Sobolev,
I posted you many mails you did not reply.
i won't waste time to send mails for MOH support every
weeks.
keep this functionality in your private repo.
Have good days
Harry
--- Maxim Sobolev <sobomax(a)portaone.com> a écrit :
> Yes, it will. I have this functionality in my
> private repo, bit
> unfortunately due to lack of free time I have not
> merged support for
> this feature into the public version of nathelper
> yet. We have some
> heavy changes that aren't applicable for the public
> version, which has
> to be filtered out.
>
> I am planning to do it sooner or later, maybe if you
> are really
> interested and can allocate some small amount of
> money to this project I
> can subcontract some of my friends/colleagues to do
> it for me.
>
> -Maxim
>
> harry gaillac wrote:
> > Hello,
> >
> > Does nathelper module + rtpproxy will be able to
> ac as
> > a 3cpp to provide MOH
> >
> > caller--------ser---------callee
> > nathelper
> > rtpproxy
> > Regards
> > Harry
> > Remarque : message transféré en pièce jointe.
> >
> >
> >
> >
> >
> >
> >
>
___________________________________________________________________________
> > Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger
> > Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> >
> >
> >
>
------------------------------------------------------------------------
> >
> > Subject:
> > Re: [Serusers] hardphones MOH
> > From:
> > harry gaillac <gaillacharry(a)yahoo.fr>
> > Date:
> > Thu, 30 Jun 2005 23:01:32 +0200 (CEST)
> > To:
> > Nils Ohlmeier <lists(a)ohlmeier.org>
> >
> > To:
> > Nils Ohlmeier <lists(a)ohlmeier.org>
> > CC:
> > serusers(a)lists.iptel.org
> >
> >
> > What do you advise me ?
> >
> > I wish to use SER+Sems(ivr) to provide MOH.
> > I've got two polycom ip300 however we just can
> place
> > on hold caller without music !
> >
> > What can I do? what's the way?
> > Does Ser or sems must act as 3cpp ?
> >
> > Regards
> > Harry
> > --- Nils Ohlmeier <lists(a)ohlmeier.org> a écrit :
> >
> >
> >>snom UA's do, except that they spare the empty
> offer
> >>to the MOH server, but
> >>send directly the SDP of the holdee to the MOH
> >>server and expecting that the
> >>server supports the chosen codec.
> >>Beware that this 3pcc MOH alternative completely
> >>breaks if there is any NAT
> >>involved.
> >>
> >> Nils
> >>
> >>On Thursday 30 June 2005 22:29, harry gaillac
> wrote:
> >>
> >>>hello,
> >>>
> >>>Anybody would be informed about ip phones able to
> >>>provide moh according to
> >>>http://www.tech-invite.com/Ti-sip-service-3.html
> >>>
> >>>regards
> >>>harry
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>
> >
>
___________________________________________________________________________
> >
> >>>Appel audio GRATUIT partout dans le monde avec le
> >>
> >>nouveau Yahoo! Messenger
> >>
> >>>Téléchargez cette version sur
> >>
> >>http://fr.messenger.yahoo.com
> >>
> >>>_______________________________________________
> >>>Serusers mailing list
> >>>serusers(a)lists.iptel.org
> >>>http://lists.iptel.org/mailman/listinfo/serusers
> >>
> >
> >
> >
> >
> >
> >
> >
> >
>
___________________________________________________________________________
>
> > Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger
> > Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hello,
I would be happy to help you however I can't help you
for setting pa module IM/presence because of I tried
to do it myself.
Try to join Jamey Hicks for help.
I can't find doc to setup pa module and xcap server.
I'm sorry !
Harry
--- meera.balu(a)wipro.com a écrit :
>
> Hi Harry
>
> I found your email id from SER development mailing
> list
>
> You seem to be testing/enhancing "pa" module of SER
>
> Iam actually testing the pa module and would need
> your help in setting
> up & doing basic testing.
>
> I have loaded the '"pa" module in ser.cfg
>
> I would like to know
>
> * Would kphone4.2 (linux based soft phone) be enough
> to test the
> "presence" aspects (Updating states, subscribing to
> users, getting
> notified of change in user states, etc.) of 'pa' ?
> * Is their any SQL table available for presence ?
> (from where did
> you get pa.sql -
>
http://lists.iptel.org/pipermail/serdev/2005-March/004129.html)
> * Other than loading the pa module in ser.cfg file,
> what are the
> other configuration changes or settings that I need
> to be doing for
> verifying presence agent module
>
> Any pointers would be of much help.
>
> Thanks
> Meera
>
>
>
>
>
>
> Confidentiality Notice
>
> The information contained in this electronic message
> and any attachments to this message are intended
> for the exclusive use of the addressee(s) and may
> contain confidential or privileged information. If
> you are not the intended recipient, please notify
> the sender at Wipro or Mailadmin(a)wipro.com
> immediately
> and destroy all copies of this message and any
attachments.
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
In fact I expect to use asterisk 1.2 for all telephony
features IM/presence, ...
I would just setup up ser + rtp proxy for far end nat
and proxying .
I agree you now when you say "i dropped sems along
time ago cause asterisk is better."
which damage sems+ser+serweb don't provide a good open
source telephony project.
My I call you via sip if possible send me you address
out of list.
regards
harry
--- Iqbal <iqbal(a)gigo.co.uk> a écrit :
> if u want some help with asterisk let me know, it
> failry simple to get
> work with ser, i dropped sems along time ago cause
> asterisk is better.
> Have voicemail and pbx, what u want to watch out for
> is how you are
> grouping callers etc, what I do is all users
> (corporates ) who need
> features of asterisk get an entry in grp table in
> ser, which says
> corporate, only these calls goto asterisk, the rest
> stay in ser, which
> saves overhead, cause asterisk does slow things
> down.
>
> Iqbal
>
> harry gaillac wrote:
>
> >Hello,
> >
> >Thanks I think I 'll do it myself when asterisk 1.2
> >will be available i will set up it like most of
> people
> >instead of sems.
> >
> >A few people are interested in sems .
> >
> >I' ll use asterisk as sip media server ser as proxy
> >and rtpproxy (far end nat) .
> >
> >Regards
> >Harry
> >
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hi Jan,
The code has been written as a SER module. The details of the code are available at:
http://developer.berlios.de/projects/osp-module
Right now the process of compilation requires that you download the OSP toolkit source code and complie it. This would build the OSP client library that the SER would use. You should then download the SER and the OSP modules and build them. The process is rather lengthy and we were wondering if we come up with an easier way of doing this.
What exactly is the experimental branch in CVS ?
Thanks,
Vikrant
---------------------------------------------------------------------------------------------------
In what form does the come come ? Did you implement it as a SER module
? If so then we can put it in the experimental branch in CVS. If you
made some changes to other modules or the core then it would be great if
you could generate patches against the latest development version.
Jan.
Sure Juan,
I am currently working on a document that details how to compile OSP with SER, and also a white paper which tells you about the benefits of using OSP with SER. It should be ready soon :) and will send it across. In the meantime if you want to learn more about OSP, you can go to www.transnexus.com.
By the way, we have also recently implemented OSP for use with the Asterisk PBX. So, if anyone here wants to use OSP for routing, security, or billing with Asterisk, let me know and I can provide more details.
Thanks,
Vikrant
----------------------------------------------------------------------
Hi Vikrant.
It seems very interesting to me. I would like you to send me the info you have.
Regards
Juan
------------------------------------------------------------
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of vmathur(a)transnexus.com
Sent: Friday, August 12, 2005 11:00 AM
To: hernan_gomez_1(a)yahoo.com
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Distributing "OSP enable SER" binary file w/ source code
Hernan,
OSP uses Public key based authentication and encrytion schemes, which are stronger than Radius's shared secrets. If you use OSP for authentication and authorization, you need not use Radius. OSP, just like Radius, has a client stack, which is implemented in the SER, and has a Server, which provides centralized call routing, accounting, and security. At the time of startup, the OSP server and the client, the SER in this case, exchange their public keys. The public and private keys can then be used to encode/decode messages as desired.
A typical call setup procedure looks like this
1. The source SER goes to the osp server to get the SIP URI corresopnding to the destination. The message is encoded using the source SER's private key.
2. The osp server decodes the message using source SERs public key and after successful decoding (authentication) returns the route back to the source. Along with the route, it also sends back a digitally signed (using the OSP Servers private key) token.
3. The Source uses the route returned by the OSP Srver to send an INVITE. The INVITE message contains the token issued by the OSP Server
4. The destination decodes/validates the token using the OSP Servers public key. Upon successful validation (authorization) it accepts the call.
This saves you the effort of mantaining cumbersome access lists for authentication. I can provide you with more documentatin on OSP and how to
use SER with OSP if you wish
Thanks,
Vikrant
-------------------------------------------------------------------------------------------------------------
Fogive my ignorance for I have never heard of OSP before ;) You mentioned Radius in your message. How does Radius authentication work in OSP? I am having a tough time getting mine to work.
hernan
vmathur(a)transnexus.com wrote:
Dear All,
I have recently implemented OSP w/ SER. OSP is an ETSI defined protocol, which I am using for ceneralised routing, and security of my inter-domain calls. The problem, however, is that the build process is a little lengthy. I want to contribute my implementation to this group so that anyone who is struggling with SER routing configurations or Radius authentication issues may benefit from it. I was, thus, wondering if we can have a binary file of the OSP enabled SER, that can be distributed with the source code. Does anyone have an opinion on this?
Also, for anyone who wants to check-out this implementation, I can provide more details.
Thanks,
Vikrant
I'm relatively new to SIP and I'm learning how to configure SER for a
little ISP.
I'm currently asking myself when we should authenticate users.
Obviously, I don't wont to have an open-relay SIP server. So I'm
thinking that I have to authenticate users for every message that comes
and that have a "From:" header that matches one of our domains.
Is this correct?
Then I have to call check_to() for REGISTER messages and check_from()
for all the others.
Is this correct?
So here it is a scheme of the logic I'm going to implement.
Do you think is correct?
IF uri == myself
IF method == REGISTER
www_authenticate()
check_to()
save()
ELSE
IF From == myself
proxy_authenticate()
check_from()
Normal processing
ELSE
IF From == myself
proxy_authenticate()
check_from()
t_relay()
ELSE
Error!
Thanks.
--
___________________________________________________
__
|- giannici(a)neomedia.it
|ederico Giannici http://www.neomedia.it
___________________________________________________
Hi,
can someone explain me, why we got a lot of context switches
while write back the memory to database?
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 30)
Looks like we have 2000 querys in that periode.
procs -----------memory---------- ---swap-- -----io---- --system--
----cpu----
r b swpd free buff cache si so bi bo in cs us sy
id wa
0 0 10716 15752 111820 793044 0 0 0 136 1292 2491 1 1
98 0
0 0 10716 15752 111820 793048 0 0 0 0 1079 1935 1 1
98 0
12 0 10716 15616 111824 793136 0 0 0 0 1896 298403 10
69 21 0
13 0 10716 15616 111824 793136 0 0 0 0 2259 400436 12
88 0 0
0 0 10716 15608 111824 793152 0 0 0 0 1691 101200 5
25 70 0
Would by write through a better solution for saving data in db?
While SER is saving the Data, some register request are timing out.
test by sipsak:
All usrloc tests completed successful.
received last message 12.012 ms after first request (test duration).
All usrloc tests completed successful.
received last message 12.069 ms after first request (test duration).
All usrloc tests completed successful.
received last message 9.171 ms after first request (test duration).
All usrloc tests completed successful.
received last message 8.777 ms after first request (test duration).
timeout after 500 ms
All usrloc tests completed successful.
received last message 6.185 ms after first request (test duration).
All usrloc tests completed successful.
received last message 7.754 ms after first request (test duration).
All usrloc tests completed successful.
received last message 7.645 ms after first request (test duration).
All usrloc tests completed successful.
received last message 10.153 ms after first request (test duration).
timeout after 500 ms
timeout after 1000 ms
version: ser 0.8.14-4 (i386/linux)
flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
@(#) $Id: main.c,v 1.168.4.3.2.1 2004/10/22 17:21:32 andrei Exp $
main.c compiled on 12:47:12 Jul 12 2005 with gcc 3.3
Greets
Markus