Hi Mike,
thanks for the answer and the tip..
I can see the sipserver sending multiple '401' to the router, so
I guess the router just drops these packets because it doesn't know
what to do with them? I now have put the UA in dmz but that still
doesn't solve the problem. I'll look further.
> -----Original Message-----
> From: Mike Tkachuk [mailto:mike@yes.net.ua]
> Sent: maandag 19 juli 2004 11:56
> To: Bart Van Daal
> Subject: Re[2]: [Serusers] NAT vs. NoNat authentication
>
>
>
> Hello Bart,
>
> Looks like UA not receiving 401 unauthorized message from
> SER, that's why it not resend REGISTER message with calculated digest.
> Maybe you have some troubles with NAT on 213.219.137.148?
>
> Hint: use ngrep with -W byline option (eg: ngrep -W byline port 5060 )
>
> --
> Best regards,
>
> ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
> Mike Tkachuk, ph:380-3433-47067
> YES ISP, fx:380-3433-47067
> Valova 17, mike|a|yes.net.ua
> Kolomyia, www.yes.net.ua
> Ukraine 78200 FWD: 66518
>
> 19.07.2004
> ICQ# 57698805
> MSN: mike_tkachuk|a|hotmail.com
> ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.
>
Hi,
I configured ser proxy, peer to peer it is working. But i want configure ser.cfg as forwarding calls to pstn gateway. dose any body help me how can i forward call from my sipsoft phone to quintum A800 gateway or any other gate through sip proxy .
plase send me sample ser.cfg
ravinder
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Thank You Andrew,
this is the ngrep output:
> -----Original Message-----
> From: Andrei Pelinescu-Onciul
> [mailto:pelinescu-onciul@fokus.fraunhofer.de]
> Sent: vrijdag 16 juli 2004 17:22
> To: Bart Van Daal
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] NAT vs. NoNat authentication
>
> On Jul 16, 2004 at 13:38, Bart Van Daal <B.Vandaal(a)edpnet.net> wrote:
> > Hi,
> >
> > Is there a difference in authenticating a natted or
> non-nated UA using
> > www_authen? The reason i'm asking is because when my UA is directly
> > connected to the internet it authenticates fine but when
> NATed I get
> > the following error:
> >
> > parse_headers: flags=4096
> > 0(12877) pre_auth(): Credentials with given realm not found
> > 0(12877) ---:: didn't authorize
> > 0(12877) build_auth_hf(): 'WWW-Authenticate: Digest
> > realm="ser.edpnet.net",
> nonce="40f7be4edbd22e214821f2a3937968fc049ae290" '
> > 0(12877) parse_headers: flags=-1
>
>
> This is normal if it happens only for the first request. Your
> UA sends the first request without auth. info., the server
> sends back a negative reply with and auth. header and then
> your UA is supposed to retry to send the request with proper auth.
>
> In the future please include network dumps.
>
> Andrei
>
Hi,
This is a new one for us. We were getting complaints of dropped calls
from one of our subscribers. We analyzed sample calls and it turns our
his SMC Router is spontaneously changing the source UDP port of the
media stream in the middle of the call!
Has anybody else seen this strange behaviour on any router?
The problem also affects SIP registrations. Even with a Keep Alive of 3
seconds in the Sipura settings, the SMC spontaneously changes ports
every 5-6 minutes.
--
Andres
Network Admin
http://www.telesip.net
Jan wrote:
Let's split the work -- you pay the license, we intergrate it.
Jan.
On 21-01 16:42, Steven R. Bunin wrote:
> Hi Guys,
>
> I have a number or users who can only do g.729 for a number of reasons
> but want voicemail. Has anyone integrated SEMS with a g.729 codec to
> record messages for missed incomming calls. If not does anyone have any
> suggestions on how to go about integrating g.729 into SEMS?
Jan,Do you have any idea of how much the license would be? Could we have a
model where is it licensed per channel? I think asterisk uses this model.
We would be willing to fund development. Aloha,Matt
Hi serusers,
I and possibly some other collegues will be in the upcoming IETF meeting.
Let me know if you would like to meet to get syncrhonized on SER, SIP, or
just go to a bar.
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hi all,
I'm sure I must be doing something very stupid! I'm using fix_nated_contact
and then t_relay(). The INVITE is being sent with a rewriten contact field,
but if the UA issues a CANCEL the private IP address is sent in the contact
field and is not rewritten.
What basic mistake have I made?!
A cut down version of my ser.cfg is:
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
fix_nated_contact();
force_rport();
if (method=="INVITE") {
fix_nated_sdp("1");
force_rtp_proxy();
/* set up reply processing */
t_on_reply("1");
setflag(6);
};
rewritehostport("213.xxx.xxx.xxx:5060");
if (!t_relay()) {
sl_reply_error();
};
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
fix_nated_contact();
if (status=~"[12][0-9][0-9]")
force_rtp_proxy();
}
Hi,
I would like to do some comprehensive logging and/or write routing scripts
based on variables.
How do I log the parts of the SIP message to a logfile ? For example how do
I log the To/From or UAC Ip address?
Is there a summary of the msg:xxxx variables somewhere (i.e. I know about
the msg:len but others?)
Thank you,
Dave.
Nope, I have down loaded latest rtpproxy and compiled on linux and
running rtpproxy without any parameters on same machine of SER.
How to enable debug on rtpproxy ?
Do I need to pass any parameters for rtpproxy ???
Do I need to set any info regarding rtp proxy in ser.cfg file ?
Please find my attached ser.cfg which is usedfor this test.
Regards,
KRC
On Fri, 9 Jul 2004 08:48:15 +0200, Andrei Pelinescu-Onciul
<pelinescu-onciul(a)fokus.fraunhofer.de> wrote:
> On Jul 08, 2004 at 17:17, Karunakar Chemudugunta <voicexml(a)gmail.com> wrote:
> [...]
> > SER is succussfully etablishing connection between two use agents and
> > but it is failed to forward rtp between two UAs. No voice.
> >
> > I assume, may be because of below errors.
> >
> > Again thanks for help.
> >
> > Regards,
> > KRC
> >
> > ERRORS
> > -----------------------------------------------------------------------------------------------
> > Jul 8 16:45:06 engcasip002 ser[18074]: ERROR: send_rtpp_command:
> > can't read reply from a RTP proxy
> > Jul 8 16:45:06 engcasip002 ser[18074]: WARNING: rtpp_test: can't get
> > version of the RTP proxy
>
>
> It looks like you don't have rtpproxy running. Start it :-)
>
>
> Andrei
>
