I think I saw somebody posting some patches to use TLS under GPL with
SER....just dig into the archives and you'll get a surprise.
Samuel.
Unclassified.
>>> "Justin Pearce" <Justin(a)PriceVideo.com> 03/24/05 11:36PM >>>
Hello all,
I am wondering if it is possible to use Kerberos with SER to get secure
user Authentication? I know that TLS would be best, however SER does not
yet impliment TLS in freely available releases.
Any thoughts on the matter? Anyone have this working in the field?
Thanks,
Justin
Hello all,
I am wondering if it is possible to use Kerberos with SER to get secure user Authentication? I know that TLS would be best, however SER does not yet impliment TLS in freely available releases.
Any thoughts on the matter? Anyone have this working in the field?
Thanks,
Justin
I have been working with CDRTool now for a couple of
days trying to get it setup. I've finnaly got SER
writing to Freeradius, etc.
Now, when I try to get in to CDRTool I get past the
login screen and am stuck with this error.
Fatal error: Cannot instantiate non-existent class:
cdrs_ser_radius in /CDRTool/callsearch.phtml on line
58
In my global.inc file I do have this class to my
knowledge. I trimmed out all of the options in the
array to the bare bones as below and still get the
error. Any ideas? The documentation is fairly light
for the tool. You would think it would be the opposite
since this is supposed to convince you to purchase a
license and get hooked.
$DATASOURCES=array(
"ser_radius" =>array(
"name" => "SIP Express Router (Radius Accounting)
",
"class" => "CDRS_ser_radius"
),
Thanks!
-J
Maybe ACKs aren't being record-routed.
Regards,
Paul
On Thu, 24 Mar 2005 10:36:02 -0300, Rodrigo P. Telles
<telles(a)devel-it.com.br> wrote:
> Hi,
>
> Java Rockx escreveu:
> > It kind of sounds like you're not sending an ACK for a 200OK response
> > from the Nextone box.
>
> Hummm, but SER should send this ACK from 200OK for everyone or is it only for
> Nextone?
> I never had this problem with other equipments.
>
> >
> > Can you send an ngrep dump of the sip call?
>
> Yes I can, I will do that.
> Thanks for your answer.
>
>
> >
> > Regards,
> > Paul
> >
> >
> > On Fri, 18 Mar 2005 13:22:04 -0300, Rodrigo P. Telles
> > <telles(a)devel-it.com.br> wrote:
> >
> >>Hi guys,
> >>
> >>I've been experiencing some troubles with carrier's that are using
> >>Nextone (www.nextone.com).
> >>I could place calls through Nextone but no longer then 1 minute, between 48 and
> >>53 seconds the call is dropped with a BYE sent by Nextone.
> >>I've been using another carriers using Cisco and Quintum without problems.
> >>Does some one experimenting this problem?
> >>
> >>I'm using SER 0.8.14.
> >>
> >>Thanks in advance.
> >>
> >>--
> >>============================================
> >>Rodrigo P. Telles <telles(a)devel.it>
> >>IVOZ # 1009
> >>Project Manager
> >>Devel-IT - http://www.devel.it
> >>Bestcom Group
> >>============================================
> >>
> >>_______________________________________________
> >>Serusers mailing list
> >>serusers(a)lists.iptel.org
> >>http://lists.iptel.org/mailman/listinfo/serusers
> >>
> >
> >
> >
>
> --
> ============================================
> Rodrigo P. Telles <telles(a)devel.it>
> IVOZ # 1009
> Gerente de Projetos
> Devel-IT - http://www.devel.it
> Grupo Bestcom
> ============================================
>
Hi there
SEMS seems to use one of the g.711 codecs by default, which uses quite a bit
of bandwidth. I believe that SEMS also ships with GSM, which uses a lot less
bandwidth. I am wondering where I would specify that I want to use this
codec instead of the default one?
Many thanks
Steven
Hi!
SER is running in my Fedora Core 3 box.
I've registered in my SER two IP phones located behind NAT, used the
example nathelper.cfg script.
How could I manage to forward calls to a PSTN gateway, example all calls
beginning by "0..." ?
What are the script modifications?
Thanks
john
Hi
I am wondering if there is some utility (either with SEMS or third party)
that can be used to monitor conferences and the users in those conferences?
Otherwise is there an alternative if want to do this?
Many thanks
Steven
Hi serusers!
After setting up an almost working ser with nat, rtpproxy and asterisk ivr,
I'm stuck on an hangup problem call if from behind nat. I can't get xlite
BYEs to hit ser (ngrep on ser machine gets nothing on hangup pressed on
client). The address in SEND>> is ser nat internal address and I'm calling
from another nat. This is xlite log on BYE
SEND TIME: 735130
SEND >> 192.168.1.100:5060
BYE sip:asterisk@192.168.1.100:5061 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.5:5060;rport;branch=z9hG4bKE406E7BA9C6E11D98395000272430644
From: gio <sip:gio@SER_PUBLIC_IP>;tag=1837372280
To: <sip:gio@SER_PUBLIC_IP>;tag=as0d4fbf68
Contact: <sip:gio@192.168.1.5:5060>
Route: <sip:192.168.1.100;ftag=1837372280;lr=on>
Call-ID: D6C66B0C-9C6E-11D9-8395-000272430644(a)192.168.1.5
CSeq: 8745 BYE
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
where 192.168.1.100 is private ip address of ser and asterisk server and
192.168.1.5 is xlite private address from another private network.
I'm pretty sure the problem is SEND >> 192.168.1.100 shoul be SEND >>
SER_PUBLIC_IP.
Reading past issues I tried fiddling with fix_rport, fix_nated_contact, but no
way. Can anyone please explain the correct way to route BYEs thru NAT?
Other things work: i can hear sounds, talk to other people and record messages
in asterisk boxes.
Please, advice
Thanx everyone
In my ser.cfg commands involved in BYE and nat are the following:
if (nat_uac_test("1")) {
if (method == "REGISTER" || !search("^Record-Route:")) {
fix_nated_contact();
if (method == "INVITE" || method == "BYE" || method == "CANCEL") {
# I think I could remove method == BYE and CANCEL with no harm...
fix_nated_sdp("1");
};
force_rport();
setflag(6);
};
};
record_route();
if (loose_route()) {
route(1);
};
if (method=="INVITE") {
if (lookup("location")) {
setflag(1);
route(1);
break;
};
};
route[1] {
if (isflagset(6)){
force_rtp_proxy();
t_on_reply("1");
};
if (method == "BYE" || method == "CANCEL"){
setflag(1);
};
if (!t_relay()){
sl_reply_error();
};
break;
}
ciao
--
Giovanni Balasso
giaso(a)yahoo.it
I'm using SER on a Public IP, do IPv4 and IPv6 Conversion, Nathelper
Module and RTPProxy. Both parties are behind NAT and gateways/routers
arranged to forward 5060 UDP and TCP ports to LAN IP's of the ATA's on
both side. There are 3 Registered Users, 8334454556 can call 8334843600,
everything is okay, each party talks, the quality of voice is great. But
when 8334843600 calls 8334454556, there's no ringing tone, nothing
happens. When i check wih serctl util, i realized one thing like ;
Contact : 'sip:8334454556@81.215.239.231:5060'
Call-ID : '13491B1111341212(a)81.215.239.231' <<< This is Public IP
Contact : 'sip:8334843600@85.96.192.140:5060'
Call-ID : '1360D1B911761212(a)192.168.0.101' <<< This is LAN IP
What may be wrong with SER or ATA's.
root@sipproxy~> serctl ul show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y
===Domain list===
---Domain---
name : 'location_inet6'
size : 512
table: 0x422b80f8
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
---Domain---
name : 'location_inet4'
size : 512
table: 0x422b6090
d_ll {
n : 3
first: 0x422ba1f8
last : 0x422ba100
}
...Record(0x422ba1f8)...
domain: 'location_inet4'
aor : '103201'
~~~Contact(0x422ba328)~~~
domain : 'location_inet4'
aor : '103201'
Contact : 'sip:103201@212.156.169.125:5060'
Expires : 1491
q : 0.00
Call-ID : '7320E5ADBA8E1526(a)212.156.169.125'
CSeq : 70
replic : 0
User-Agent: 'Unknown'
State : CS_NEW
Flags : 0
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
...Record(0x422ba2e0)...
domain: 'location_inet4'
aor : '8334454556'
~~~Contact(0x422ba238)~~~
domain : 'location_inet4'
aor : '8334454556'
Contact : 'sip:8334454556@81.215.239.231:5060'
Expires : 56
q : 0.00
Call-ID : '13491B1111341212(a)81.215.239.231'
CSeq : 128
replic : 0
User-Agent: 'Unknown'
State : CS_NEW
Flags : 0
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
...Record(0x422ba100)...
domain: 'location_inet4'
aor : '8334843600'
~~~Contact(0x422ba148)~~~
domain : 'location_inet4'
aor : '8334843600'
Contact : 'sip:8334843600@85.96.192.140:5060'
Expires : 32
q : 0.00
Call-ID : '1360D1B911761212(a)192.168.0.101'
CSeq : 8
replic : 0
User-Agent: 'Unknown'
State : CS_NEW
Flags : 0
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
root@sipproxy~>
--
>>> please do not use
>>> l.s.d(a)dnaofdesign.com
>>> above account is stand by
>>> use same account as
>>> cosmocid(a)ispro.net.tr
How can we restrict the registration to only one user per login?.
Currently, you can register with the same login/password in different
machines at the same time. But what we require is that only the first
registration is valid and the rest ones are invalidated by the SER.
Thanks
