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" <nolivier(a)alphalink.fr>
Subject: Re: [Users] nat_helper: multiple media IP address in SDP
To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro>
Cc: users(a)openser.org
Message-ID: <443BC26A.8040407(a)alphalink.fr>
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:0143132445@192.168.1.20>;tag=as226ce7b9.
> > To: <sip:0123456789@192.168.1.20:5062>;tag=3123AAA8-20C5.
> > Date: Tue, 11 Apr 2006 09:10:29 GMT.
> > Call-ID: 079ab6663e403ff44a1107e5111b075f(a)192.168.1.20.
> > Server: Cisco-SIPGateway/IOS-12.x.
> > CSeq: 102 INVITE.
> > Allow-Events: telephone-event.
> > Contact: <sip:677238#0123456789@172.16.0.10:5060>.
> > 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:0143132445@192.168.1.20>;tag=as1a2f900d.
> > To: <sip:0123456789@192.168.1.20:5062>;tag=3121D1B4-1BFD.
> > Date: Tue, 11 Apr 2006 09:08:28 GMT.
> > Call-ID: 08467c5e299ab833106517c63d3edc2e(a)192.168.1.20.
> > Server: Cisco-SIPGateway/IOS-12.x.
> > CSeq: 102 INVITE.
> > Allow-Events: telephone-event.
> > Contact: <sip:677238#0123456789@172.16.0.10:5060>.
> > 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
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------------------------------
Message: 7
Date: Tue, 11 Apr 2006 16:52:14 +0200
From: Cesc <cesc.santa(a)gmail.com>
Subject: Re: [Users] Allow only TLS connections
To: "Thorsten.Haupt(a)t-systems.com" <Thorsten.Haupt(a)t-systems.com>
Cc: users(a)openser.org
Message-ID:
<ce8208420604110752i733143d8k5b565ac45b0cfda2(a)mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook&DokuWiki=6
c17b007ea61fa37b86b391ce1b2a80f#tcp
On 4/11/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com>
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:cesc.santa@gmail.com]
Gesendet: Dienstag, 11. April 2006 14:03
An: Haupt, Thorsten
Cc: users(a)openser.org
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, Thorsten.Haupt(a)t-systems.com
<Thorsten.Haupt(a)t-systems.com>
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
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------------------------------
Message: 8
Date: Tue, 11 Apr 2006 17:16:49 +0200
From: Andreas Granig <andreas.granig(a)inode.info>
Subject: [Users] Overlapping AVPs
To: users(a)openser.org
Message-ID: <443BC861.6040303(a)inode.info>
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
$avp(i:102)[1] maybe?
Thanks,
Andy
------------------------------
Message: 9
Date: Tue, 11 Apr 2006 17:54:56 +0200
From: "D'Addelfio Davide" <Davide.D'Addelfio(a)italtel.it>
Subject: R: [Users] load from db table
To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro>
Cc: users(a)openser.org
Message-ID: <82C94EFCF026F74EB91A2048B1C963A504065277(a)BESONE.corp.dom>
Content-Type: text/plain; charset="iso-8859-1"
Hi Bogdan, I'm trying to setup my config file , did the same
modparam("avpops", "db_scheme",
"scheme0:username_col=from;value_col=timestamp;value_type=string;table=a
cc")
modparam("avpops","avp_aliases","timestamp=i:800")
if (method=="INVITE")
avp_db_load("$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: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per
conto di Bogdan-Andrei Iancu
Inviato: martedм 11 aprile 2006 16.12
A: D'Addelfio Davide
Cc: users(a)openser.org
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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End of Users Digest, Vol 11, Issue 27
*************************************