I have been unable to download SER from ftp://ftp.berlios.de/pub/ser/ <ftp://ftp.berlios.de/pub/ser/> for several days. Several colleagues have reported the same problem. The ftp server appears to be "busy" at all times even though it is never near its maximum of 95 users.
I have downloaded SER in the past for several projects without problems, so this seems to be a relatively recent change.
Are there mirror sites from which to download a SER installation package? I found source packages on http://developer.berlios.de/projects/ser/ <http://developer.berlios.de/projects/ser/> . Are pre-built executable SER packages no longer distributed?
Thanks.
Doug Davey
On 17-11-2005 10:21, Lenir wrote:
> In this case the radius proxy wont work, because you never can anticipate
> who hangs up the call, thus radius wont know who hung up the call...Besides
> all other voice applications/hardware (SIP and H323) that use radius do not
> behave like that, the Called-Station-ID ALWAYS remains the same, as with the
> Calling-Station-ID.
Could you name those SIP applications that behave the way you describe ?
Jan.
Hi,
I have implemented a new group feature. So far we have group support
with fixed matching: user X belongs to group G, etc..
This proves to be inefficient when you have huge block of users
belonging to small numbers of groups. So I did some regular expression
based groups:
if user@domain matched a RE -> belongs to group G
very usefull if try to provisions routing logic based on groups.
I developed this new functionality inside the group module since there
is about same functionality and it's too small to be a separate module.
any pro or against comments / suggestions before committing?
regards,
Bogdan
Hi messa!
I suggest you to take a look on the getting_started document from
onsip.org. It uses mediaproxy module and the mediaproxy (just another
flavour of nathelper+rtpproxy).
You can't integrate only nathelper to vocal. Use openser+nathelper as
replacement for vocal or together with vocal.
btw: you could resolve the problem by using clients which support ICE.
regards
klaus
PS: please no direct email. Send your emails always to the list.
messa(a)innsof.com wrote:
> Hi Klaus
>
> I want you to help me. I want to integrate nathelper in my vocal
> system to
> fix the problem of NAT traversal because the STUN solution is not good for
> me. With stun, i cannot make in the same moment a call to another client
> in a remote network and to a client in the same network. ANd NAthelper do
> it good for openser. So i want to know, the architecture of openser. How
> call the nathelper module. I want to have the architecture of this
> nathelper module.
>
> Thanks in advance!
>
> Best regards
>
> Serge
>
>
messa(a)innsof.com wrote:
> Hi Klaus
>
> I want you to help me. I want to integrate nathelper in my vocal
> system to
> fix the problem of NAT traversal because the STUN solution is not good for
> me. With stun, i cannot make in the same moment a call to another client
> in a remote network and to a client in the same network. ANd NAthelper do
> it good for openser. So i want to know, the architecture of openser. How
> call the nathelper module. I want to have the architecture of this
> nathelper module.
>
> Thanks in advance!
>
> Best regards
>
> Serge
>
>
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep the sessions via SER .
Ser receive packets with private ip in contact field
which one is forward to asterisk .
How ser can handle the contact field to establish sip
sessions between sip agents and asterisk ?
I've been trying mangle and textops modules but i
really need to be advice.
One box
---------------------------
| ---------------- |
| | asterisk pbx | |
| ---------------- |
| || |
| ---------- ----------
| | SER |====|NAT box |==== private network
| ---------- ----------
---------------------------
Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hello,
Here is my config :
Asterisk as registrar server :public ip:5050
Ser as outbound proxy server :public ip 5060
I wish ser to handle the packets between Nat box
(netfilter) and Asterisk However contact field in
the sip HF don't change from nat box to asterisk which
don't allow to keep the sessions via SER .
Ser receive packets with private ip in contact field
which one is forward to asterisk .
How ser can handle the contact field to establish sip
sessions between sip agents and asterisk ?
I've been trying mangle and textops modules but i
really need to be advice.
One box
---------------------------
| ---------------- |
| | asterisk pbx | |
| ---------------- |
| || |
| ---------- ----------
| | SER |====|NAT box |==== private network
| ---------- ----------
---------------------------
Regards
Harry
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
hi all
i donot know howto set the port of RTPproxy, my server public ip such
as 202.97.123.123, i want to set RTPproxy use the 5080-5082 ports,what can i
do for it. client can use 202.97.123.123:5080-5082 as a outbound proxy!
Hello Nils,
Attached is the mentioned files you asked me to send. As you said, after
disabling gnutls in configure, I can compile it. It works fine . Thanks for
your kind help.
One important question I would like to ask you.
Suppose I am having 100 SIP clients, all of them are behind their respective
100 NAT's, The NAT devices are assigned public IP dynamically. What besides
openser do I need on my SIP server to traverse those NAT's. Assume NAT's
does not support STUN. Do I need to put STUN or Mediaproxy/RTPProxy server
on my SIP server. If not , then how could i can manage such network.
Thanks a lot for your kind help
Best Regards
Lokesh
----- Original Message -----
From: "Nils Ohlmeier" <lists(a)ohlmeier.org>
To: "Lokesh Kumar" <lokesh(a)interacesso.pt>
Sent: Thursday, November 17, 2005 10:14 PM
Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT
> Hello Lokesh,
>
> could you do me a favor and send me config.h and config.log from the
directory
> where you tried to compile 0.9.2?
>
> After you send me the files you could try the following command
> './configure --disable-gnutls' and afterwards 'make'. I think that should
help
> to compile sipsak-0.9.2 on your system.
>
> Thanks
> Nils
>
> On Friday 18 November 2005 19:54, you wrote:
> > Hello Nils,
> >
> > When I do make in the source directory of sipsak-0.9.2, it gives me
error
> > like this.
> >
> > /usr/lib/libgnutls-openssl.so: undefined reference to
> > 'asn1_delete_structue' /usr/lib/libgnutls-openssl.so: undefined
reference
> > to 'asn1_create_element' /usr/lib/libgnutls-openssl.so: undefined
reference
> > to 'asn1_der_decoding' collect2: ld returned 1 exit status
> > make[1]: ***[spsak] Error 1
> > make[1]: Leaving directory '/usr/src/sipsak-0.9.2'
> > make: *** [all] Error 2
> >
> > But there is no such problem with 0.9.0 version.
> >
> > Regards
> >
> > Lokesh
> >
> >
> > ----- Original Message -----
> > From: "Nils Ohlmeier" <lists(a)ohlmeier.org>
> > To: <serdev(a)iptel.org>
> > Cc: "Lokesh Kumar" <lokesh(a)interacesso.pt>; <deepak.dhiman(a)uf4.net>;
> > <serusers(a)iptel.org>
> > Sent: Thursday, November 17, 2005 6:05 PM
> > Subject: Re: [Serdev] How to gaurd PSTN gateway with openser + NAT
> >
> > > Hello Lokesh,
> > >
> > > On Friday 18 November 2005 01:27, Lokesh Kumar wrote:
> > > > I installed openser, it is working fine. sipsak-0.9.0 is also
working
> >
> > fine,
> >
> > > > previously i was using 0.9.2 version of sipsak that is having
problem.
> > >
> > > I would really appreciate it if you could describe we what problems
you
> >
> > had
> >
> > > with sipsak 0.9.2. Then I could try to fix it, if it is a bug in
sipsak.
> > >
> > > Thanks
> > > Nils
Hi,
thanks to Norman Brandinger we have a basic tutorial about OpenSER
scripting. The document is linked in doc section on the project page,
along with the other tutorials:
http://www.openser.org/docs
also you can accessing directly at:
http://www.openser.org/docs/scripting.html
Again, many thanks to Norman for his contribution.
Best regards,
Bogdan
