Hi Dmitry,

Thanks, it works like a charm.

Nicolas

Dmitry Lyubimkov wrote:
We had the same problem with SDP.
There are very many UA with this mistake error
To bypass this restriction we have added after
force_rtp_proxy();
Also such command
subst("/^c=IN IP4 ([0-9]+)\.([0-9]+)\.([0-9]+)\.([0-9]+)(.*)/c=IN IP4
11.22.33.44\5/i");

Dmitry

------------------------------

Message: 6
Date: Tue, 11 Apr 2006 16:51:22 +0200
From: "Nicolas Olivier" <[EMAIL PROTECTED]>
Subject: Re: [Users] nat_helper: multiple media IP address in SDP
To: "Bogdan-Andrei Iancu" <[EMAIL PROTECTED]>
Cc: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;       format=flowed;  charset="ISO-8859-1"



Ok, I may have a look to the csv.
Thanks for the help.

regards,
Nicolas

Bogdan-Andrei Iancu wrote:
 > Hi,
 >
 > Nicolas Olivier wrote:
 >
 >  >
 >  > Hi Bogdan,
 >  >
 >  > Ok, I understand now. But I still encounter the problem because:
 >  > - rtpproxy only rewrites the c= from media part (but it should be
fine
 >  > as you said) despite what a quick look in the rtpproxy code
comments
 >  > say ("We have to change ports in m-lines, and also change IP
addresses
 >  > in c-lines which can be placed either in session header (fallback
for
 >  > all medias) or media description.")
 >
 > yes, the nathelper will change the c= from session header only if it
 > finds a media section without a local c= (which means the default c=
 > from session hdr will be used).
 >
 >  > - the centrex (which is an asterisk by the way) take only into
account
 >  > the c= from the session part, not the one from media part
 >
 > in the CVS devel there is a flag that force also changing of session
c= :
 >     http://openser.org/docs/modules/1.1.x/nathelper.html#AEN275 , the
 > "c" flag
 >
 > regards,
 > bogdan
 >
 >  >
 >  >
 >  >
 >  > regards,
 >  > Nicolas
 >  >
 >  > Bogdan-Andrei Iancu wrote:
 >  >
 >  >> Hi Nicolas,
 >  >>
 >  >> it;s perfectly ok - see the SDP RFC : an SDP may contain a default

 > c= in
 >  >> the session part; each media section (m=) may contain an ip (c=);
if it
 >  >> doesn't the session c= will be used.
 >  >>
 >  >> regards,
 >  >> bogdan
 >  >>
 >  >> Nicolas Olivier wrote:
 >  >>
 >  >>  >
 >  >>  > Hi,
 >  >>  >
 >  >>  > I've got a gateway which is only used for rounting and rtp
proxying
 >  >>  > between providers and centrexes.
 >  >>  >
 >  >>  > On reply to an INVITE, one of our provider send back a "183
Session
 >  >>  > Progress". The problem is that in the SDP block, we've got 2
 > media IP
 >  >>  > address and rtpproxy only rewrite one.
 >  >>  >
 >  >>  > Finally, the provider establish rtp session with our gateway,
and
 > our
 >  >>  > centrex directly with the provider.
 >  >>  >
 >  >>  >   provider                  gateway                  centrex
 >  >>  > 172.16.0.10               192.168.1.10
192.168.1.20
 >  >>  >      RTP     ------------->   RTP      ------------>   RTP
 >  >>  >       ^-------------------------------------------------|
 >  >>  >
 >  >>  > So my questions are, is it possible to have multiple IP address
in
 >  >> SDP
 >  >>  > and if so, how can I tell rtpproxy to rewrite all of them.
 >  >>  >
 >  >>  > Coming from provider:
 >  >>  >
 >  >>  > SIP/2.0 183 Session Progress.
 >  >>  > Via: SIP/2.0/UDP
 >  >>  > 192.168.1.10;branch=z9hG4bKdd67.a4cc2c44.0,SIP/2.0/UDP
 >  >>  > 192.168.1.20:5062;branch=z9hG4bKdd67.08f45a33.0,SIP/2.0/UDP
 >  >>  > 192.168.1.20:5060;branch=z9hG4bK4af242b7.
 >  >>  > From: "02" <sip:[EMAIL PROTECTED]>;tag=as226ce7b9.
 >  >>  > To: <sip:[EMAIL PROTECTED]:5062>;tag=3123AAA8-20C5.
 >  >>  > Date: Tue, 11 Apr 2006 09:10:29 GMT.
 >  >>  > Call-ID: [EMAIL PROTECTED]
 >  >>  > Server: Cisco-SIPGateway/IOS-12.x.
 >  >>  > CSeq: 102 INVITE.
 >  >>  > Allow-Events: telephone-event.
 >  >>  > Contact: <sip:[EMAIL PROTECTED]:5060>.
 >  >>  > Record-Route:
 >  >>  >
 >  >>
 >
<sip:192.168.1.10;ftag=as226ce7b9;lr=on>,<sip:192.168.1.20:5062;ftag=as2
26ce7b9;lr=on>.
 >
 >  >>
 >  >>  >
 >  >>  > Content-Disposition: session;handling=required.
 >  >>  > Content-Type: application/sdp.
 >  >>  > Content-Length: 261.
 >  >>  > .
 >  >>  > v=0.
 >  >>  > o=CiscoSystemsSIP-GW-UserAgent 3448 4768 IN IP4 172.16.0.10.
 >  >>  > s=SIP Call.
 >  >>  > c=IN IP4 172.16.0.10.
 >  >>  > t=0 0.
 >  >>  > m=audio 18322 RTP/AVP 18 101.
 >  >>  > c=IN IP4 172.16.0.10.
 >  >>  > a=rtpmap:18 G729/8000.
 >  >>  > a=fmtp:18 annexb=no.
 >  >>  > a=rtpmap:101 telephone-event/8000.
 >  >>  > a=fmtp:101 0-16.
 >  >>  >
 >  >>  > Forwarded to centrex:
 >  >>  >
 >  >>  > SIP/2.0 183 Session Progress.
 >  >>  > Via: SIP/2.0/UDP
 >  >>  > 192.168.1.20:5062;branch=z9hG4bK43a4.3e96aba3.0,SIP/2.0/UDP
 >  >>  > 192.168.1.20:5060;branch=z9hG4bK3213db83.
 >  >>  > From: "02" <sip:[EMAIL PROTECTED]>;tag=as1a2f900d.
 >  >>  > To: <sip:[EMAIL PROTECTED]:5062>;tag=3121D1B4-1BFD.
 >  >>  > Date: Tue, 11 Apr 2006 09:08:28 GMT.
 >  >>  > Call-ID: [EMAIL PROTECTED]
 >  >>  > Server: Cisco-SIPGateway/IOS-12.x.
 >  >>  > CSeq: 102 INVITE.
 >  >>  > Allow-Events: telephone-event.
 >  >>  > Contact: <sip:[EMAIL PROTECTED]:5060>.
 >  >>  > Record-Route:
 >  >>  >
 >  >>
 >
<sip:192.168.1.10;ftag=as1a2f900d;lr=on>,<sip:192.168.1.20:5062;ftag=as1
a2f900d;lr=on>.
 >
 >  >>
 >  >>  >
 >  >>  > Content-Disposition: session;handling=required.
 >  >>  > Content-Type: application/sdp.
 >  >>  > Content-Length: 277.
 >  >>  > .
 >  >>  > v=0.
 >  >>  > o=CiscoSystemsSIP-GW-UserAgent 565 174 IN IP4 172.16.0.10.
 >  >>  > s=SIP Call.
 >  >>  > c=IN IP4 172.16.0.10.
 >  >>  > t=0 0.
 >  >>  > m=audio 36296 RTP/AVP 18 101.
 >  >>  > c=IN IP4 192.168.1.10.
 >  >>  > a=rtpmap:18 G729/8000.
 >  >>  > a=fmtp:18 annexb=no.
 >  >>  > a=rtpmap:101 telephone-event/8000.
 >  >>  > a=fmtp:101 0-16.
 >  >>  > a=nortpproxy:yes.
 >  >>  >
 >  >>  >
 >  >>  > openser.cfg
 >  >>  >
 >  >>  > (...)
 >  >>  >
 >  >>  >  onreply_route[1] {
 >  >>  >          if (status =~ "(180)|(183)|2[0-9][0-9]") {
 >  >>  >                  fix_nated_contact();
 >  >>  >                  if (!search("^Content-Length:[ ]*0")) {
 >  >>  >                          force_rtp_proxy();
 >  >>  >                  };
 >  >>  >          } else if (nat_uac_test("1")) {
 >  >>  >                  fix_nated_contact();
 >  >>  >          };
 >  >>  >  }
 >  >>  >
 >  >>  > (...)
 >  >>  >
 >  >>  > Best regards,
 >  >>  > Nicolas Olivier
 >  >>  >
 >  >>  >
 >  >>  > _______________________________________________
 >  >>  > Users mailing list
 >  >>  > [email protected]
 >  >>  > http://openser.org/cgi-bin/mailman/listinfo/users
 >  >>  >
 >  >>
 >  >
 >




------------------------------

Message: 7
Date: Tue, 11 Apr 2006 16:52:14 +0200
From: Cesc <[EMAIL PROTECTED]>
Subject: Re: [Users] Allow only TLS connections
To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Cc: [email protected]
Message-ID:
        <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1

http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6 <http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6>
c17b007ea61fa37b86b391ce1b2a80f#tcp



On 4/11/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]>
wrote:
 > I searched for this function, but I didn't found it :-(
 > Knows anyone the correct code, not only pseudo-code?
 >
 > Torsten
 >
 > -----Ursprьngliche Nachricht-----
 > Von: Cesc [mailto:[EMAIL PROTECTED]
 > Gesendet: Dienstag, 11. April 2006 14:03
 > An: Haupt, Thorsten
 > Cc: [email protected]
 > Betreff: Re: [Users] Allow only TLS connections
 >
 > I think in openser there is a function to check what transport the
message came in ... you can do something like:
 > if ( transport != TLS ) {
 >           send error to UA
 >           break;
 > }
 >
 > Cesc
 >
 > On 4/11/06, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
 > >
 > >
 > > Hello,
 > >
 > > I use OpenSER in a testing environment for VoIP security. My clients
 > > connect via TLS. If I deactivate UDP/5060 on the server, it doesn't
work correct.
 > > Some Clients can't connect and others can't establish calls. I read
in
 > > another thread, that UDP is mandatory for SIP and that the server
need it.
 > >
 > > But how can I prevent users from connecting via UDP and force them
to
 > > use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But
is
 > > this the correct way? Are there any parameters server-side to force
 > > users to connect via TLS?
 > >
 > > Thanks for response.
 > > Torsten
 > > _______________________________________________
 > > Users mailing list
 > > [email protected]
 > > http://openser.org/cgi-bin/mailman/listinfo/users
 > >
 > >
 > >
 >
 > _______________________________________________
 > Users mailing list
 > [email protected]
 > http://openser.org/cgi-bin/mailman/listinfo/users
 >



------------------------------

Message: 8
Date: Tue, 11 Apr 2006 17:16:49 +0200
From: Andreas Granig <[EMAIL PROTECTED]>
Subject: [Users] Overlapping AVPs
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi,

Me again, sorry, but the docs aren't really noisy about AVP details...

So if I have user preferences for both the caller and callee and load
them from DB and print them using the following:

   avp_db_load("$avp($uuid_caller)", "");
   avp_db_load("$avp($uuid_callee)", "");
   avp_print();

then they may overlap because of the same ID (say "i:102" for toggling
some specific feature on/off), but according to the debug output both
are present:

   INFO:avpops:print_avp: p=0x4056db90, flags=100
   INFO:                   id=<102>
   INFO:                   val_int=<1>

   INFO:avpops:print_avp: p=0x4056dc68, flags=100
   INFO:                   id=<102>
   INFO:                   val_int=<0>

So is it possible to selectively access the avp-value of both
$uuid_caller and $uuid_callee? Something like $avp(i:102)[0] and
$avp(i:102)[1] maybe?

Thanks,
Andy



------------------------------

Message: 9
Date: Tue, 11 Apr 2006 17:54:56 +0200
From: "D'Addelfio Davide" <Davide.D'[EMAIL PROTECTED]>
Subject: R: [Users] load from db table
To: "Bogdan-Andrei Iancu" <[EMAIL PROTECTED]>
Cc: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;       charset="iso-8859-1"

Hi Bogdan, I'm trying to setup my config file , did the same

modparam("avpops", "db_scheme",
"scheme0:username_col=from;value_col=timestamp;value_type=string;table=a
cc")
modparam("avpops","avp_aliases","timestamp=i:800")

if (method=="INVITE")
avp_db_load("$from","$timestamp/$scheme0");

I'm not sure that is correct...

log gives me these errors

Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: submit_query: You
have an error in your SQL syntax.  Check the manual that corresponds to
your MySQL server version for the right syntax to use near 'from='bob''
at line 1
Apr 11 08:42:59 localhost /usr/sbin/openser[13752]: db_query: Error
while submitting query
Apr 11 08:42:59 localhost /usr/sbin/openser[13752]:
ERROR:avpops:load_avps: db_load failed

Any thought?

Thanks
Davide

-----Messaggio originale-----
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per
conto di Bogdan-Andrei Iancu
Inviato: martedм 11 aprile 2006 16.12
A: D'Addelfio Davide
Cc: [email protected]
Oggetto: Re: [Users] load from db table

Hi,

see:
    http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load

the "db_scheme" example.

regards,
bogdan

D'Addelfio Davide wrote:

 >Hi Bogdan,
 >
 >i setup my openser.cfg to store SIP messages into acc tables of mysql,
 >using extra accounting to store also the body part of the message.
 >Now i need that openser read into that db's table, in particular in
some
 >rows of db.
 >If I use avp_db_load it works only over usr_preference table, instead I
 >want it looks into acc table.
 >
 >How can I do?
 >
 >Thanks for help
 >Davide
> >


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------------------------------

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End of Users Digest, Vol 11, Issue 27
*************************************





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