Mahesh,
I now know what the problem is, but I don't have any idea the cause or
how to fix it. Fortunately the issue appears to be our PSTN gateway
provider.
See we use a 3rd party for PSTN access. This 3rd party uses a
[unknown] SIP proxy in front of their Sonus box.
The problem I have with them is that occasionally their SIP proxy
changes the branch= tag in the top VIA which then appears to be a new
re-INVITE and I believe SER properly processes it. However, the SIP UA
always ignores it and I don't know if it should or not.
Anyhow here is a sample of what I'm referring to. The first re-INVITE
is OK and is properly processed however we get a seconds re-INVITE
from them (also shown) and this second re-INVITE is received
__before__ we receive the ACK for our 200OK which we send back to them
in response to the first re-INVITE.
You can see the top VIA is different.
FIRST RE-INVITE
U 2005/03/22 18:17:34.949888 216.229.127.80:5060 -> 10.3.0.221:5060
INVITE sip:7246024356@24.154.237.253:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.229.127.80:5060;branch=z9hG4bK8152abf9bd6-899f6309.
Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bK07650ade1b9163d1.
To: "Paul" <sip:7246024356@216.229.127.80>;tag=4217611370.
From: sip:4075660914@sip.mycompany.com;tag=0a8eb95c.
Call-ID: 2063115359(a)24.154.237.253.
CSeq: 15749 INVITE.
Max-Forwards: 69.
Contact: sip:4075660914@216.229.118.76:4060.
Record-Route: <sip:216.229.127.80:5060;lr>.
Route: <sip:10.3.0.221:5060;ftag=4217611370;lr>.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Accept: multipart/mixed, application/sdp, application/isup,
application/dtmf, application/dtmf-relay.
Supported: timer.
Session-Expires: 240;refresher=uac.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 249.
.
v=0.
o=Sonus_UAC 10885 11051 IN IP4 216.229.118.76.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 19432 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
SECOND RE-INVITE
U 2005/03/22 18:17:35.450645 216.229.127.80:5060 -> 10.3.0.221:5060
INVITE sip:7246024356@24.154.237.253:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.229.127.80:5060;branch=z9hG4bK9157a393106-899f6309.
Via: SIP/2.0/UDP 216.229.118.76:4060;branch=z9hG4bK07650ade1b9163d1.
To: "Paul" <sip:7246024356@216.229.127.80>;tag=4217611370.
From: sip:4075660914@sip.mycompany.com;tag=0a8eb95c.
Call-ID: 2063115359(a)24.154.237.253.
CSeq: 15749 INVITE.
Max-Forwards: 69.
Contact: sip:4075660914@216.229.118.76:4060.
Record-Route: <sip:216.229.127.80:5060;lr>.
Route: <sip:10.3.0.221:5060;ftag=4217611370;lr>.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Accept: multipart/mixed, application/sdp, application/isup,
application/dtmf, application/dtmf-relay.
Supported: timer.
Session-Expires: 240;refresher=uac.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 249.
.
v=0.
o=Sonus_UAC 10885 11051 IN IP4 216.229.118.76.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 19432 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
I can post a complete example SIP call log which shows this if you
would like to look at it.
Also, I wonder if the SIP UA ignores the "second" re-INVITE because it
is in a transaction with the first re-INVITE (because it didn't get
the ACK back from the PSTN GW provider yet).
I'm not familiar enough with the RFC to know.
The bottom line is that last night I finally got our PSTN GW provider
to admit they have something going on. They're looking in to why the
top VIA is branching. No solution yet.
Regards,
Paul
On Wed, 23 Mar 2005 07:50:37 -0600, Mahesh Subramanya <mahesh(a)aptela.com> wrote:
> We're dealing with (somewhat) the same issue, interestingly, involving
> Sonus too. On our end, I've noticed that different UAs deal better (or
> worse) with this issue - Snoms seem to have no problem responding
> repeatedly to the same INVITE (whether they should or not is a different
> issue :-) ), while Polycoms just tend to crap out.
>
> Anyhow, did you figure out a "correct" soln?
>
> cheers
>
>
Trying to make the auth_radius module to work I ran into a peculiar issue.
For example if our UA were to try to register to server "sip.mydomain.com"
...and our ser.cfg had:
if (!radius_www_authorize("mydomain.com")) {
www_challenge("mydomain.com", "1");
}
...then the authentication is not even fired off to the radius. SER
Debugs indicate the radius message is not even constructed.
If on the other hand our ser.cfg has:
if (!radius_www_authorize("")) {
www_challenge("", "1");
}
then the authentication is now fired off to the radius server but the
REALM is sip.mydomain.com.
Why can't one make this work as it does with mysql authentication where
the www_authorize does not need the host part? We need REAM to be
simply the domain part.
The auth_radius readme even says that the realm is **usually** just the
domain of the host. Does this mean something is broken here?
-------------from readme----------------
* realm - Realm is a opaque string that the user agent
should present to the user so he can decide what username
and password to use. Usually this is domain of the host
the server is running on.
Example 1-3. radius_www_authorize usage
...
if (!radius_www_authorize("iptel.org")) {
www_challenge("iptel.org", "1");
};
--
Andres
Network Admin
http://www.telesip.net
Check /var/log/messages
Assuming a standard Linux distro.
Regards,
Paul
On Wed, 23 Mar 2005 04:19:24 -0800 (PST), Kamran Ahmad <p_kami(a)yahoo.com> wrote:
> hi Java Rockx
>
> do you know when
>
> #ser.cfg
> route {
> log(1,"some message");
> }
>
> how to find this message
> in which file ser is filling its log
>
> thanks
> Kamran
>
>
> --- Java Rockx <javarockx(a)gmail.com> wrote:
> > The only way I know is to start up gdb and set some
> > tracepoints
> >
> > Regards,
> > Paul
> >
> >
> > On Wed, 23 Mar 2005 02:33:25 -0800 (PST), Kamran
> > Ahmad <p_kami(a)yahoo.com> wrote:
> > >
> > > hi all
> > >
> > > i want to check that if my ser is calling radius
> > for
> > > authentication. is this posible in ser
> > >
> > > Kamran
> > >
> > > __________________________________
> > > Do you Yahoo!?
> > > Yahoo! Sports - Sign up for Fantasy Baseball.
> > > http://baseball.fantasysports.yahoo.com/
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> > >
> >
>
> __________________________________
> Do you Yahoo!?
> Make Yahoo! your home page
> http://www.yahoo.com/r/hs
>
Hi Jiang,
Be sure you record_route() also to calls forwarded to GW - add the
record_route() before the forward().
Also don't forget to do loose_route() in your script for requests within
the dialog.
Best regards,
Marian
Jiang zhou wrote:
> Hi,all
>
> I have a record route problem:
>
> sip client===>ser====>(pstn)gateway
>
> invite: client---->ser---->gw
> ring and 200ok : gw--->ser--->client
> ack : client--------------->gw
>
> I use forward command forward pstn call to gateway.
> if (uri=~"^sip:9817") {
> rewritehost("xxx.xxx.xxx.xxx");
> forward(xxx.xxx.xxx.xxx,5060);
> break;
> }
>
> There is no record-route field in the 200ok packet. So client forward to gw
> directly.
> But when sip client call each other ser insert record-route field. Can
> someone tell me
> how to config forward command with record-route supported?
>
> Jiangzhou
> Best Regards
>
>
>
--
Voice System
http://www.voice-system.ro
Dear All,
Paul, Greger and myself would like you to know that the next issue of
'Getting Started' is now available on the www.ONsip.org web site.
This new version builds upon the first issue by adding MySQL support.
We are already finalising the next issue covering NAT support - a
subject that always fills up this mailing list with questions.
We are doing this for the community and would welcome any feedback on
the document to make it more readable etc. Comments are welcome on the
<http://www.onsip.org/> www.ONsip.org web site, goto Forums and select
the Getting Started Feedback area.
To download the document, go to the download section.
We hope this is of benefit to the Community.
Thanks
Paul, Simon, Greger
Hi Uli,
Please Help. I still have errors in the startup of the sems. I have tried
the ATDT <Telephone> and I get the NO DialTone and No MSN & EAZ error.
Please advice what can be the problem as the system already detected the
card but there is not response. Thanks in advance.
regards,
nicky
Hello,
I've discovered recently the VoIP world. I would like to install SER in
Fedora Core 3
I think the latest SER stable version is 0.8.14 and does not have a RPM
install package.
If you succeeded making SER to work in FC3 could you tell me what is the
"best practice" to do it?
thanks,
luis
Hi all !
I use CDRTool for accounting with radiusclient 0.4.3 librairy and freeradius
1.0.2.
For Start and Stop status code, I have no problem, I have the CDRs on MySQL
Database.
But when I want to have MISSED CALL, I have an error on freeradius !
Error :
Mon Mar 21 17:10:53 2005 : Error: rlm_radutmp: NAS localhost port 5060
unknown packet type 15)
Mon Mar 21 17:10:53 2005 : Info: rlm_sql (sql): Unsupported Acct-Status-Type
= 15
Does someone know how to solve this problem??
Is it in the SQL.CONF ??
Thanks a lot
Vos Solutions Voix-Data !
Nicolas Ruiz
Service Technique
Ligne directe : + 33 (0) 1 56 38 39 71
Fax :+ 33 (0) 1 47 24 74 77
nruiz(a)vivaction.com
Immeuble Plein Ouest
177 av. Georges Clemenceau
92024 Nanterre - France
Tel : 0 811 02 6000
www.vivaction.com
____________________________________________________________________________
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Hi All.
I'm using ser-0.9.1
If I store a value using avp_write() how long does this AVP exist?
Do I need to explicitly call avp_delete() to free memory resources or
will they expire and clean themselves up automatically?
Regards,
Paul
Marian
Thanks for your answer...
I use netstan -an, but how can i determine the PRIMARY BIND ADDRESS and the
ALTERNATIVE BIND ADDRESS..which is which?
Thanks..
Regards,
Ricardo.-
> -----Mensaje original-----
> De: Marian Dumitru [mailto:marian.dumitru@voice-sistem.ro]
> Enviado el: Martes, 22 de Marzo de 2005 14:53
> Para: Ricardo Martinez
> CC: 'serusers(a)lists.iptel.org'
> Asunto: Re: [Serusers] Question about mystun.
>
>
> Hi Ricardo,
>
> Use "netstat -ual" to the interfaces.
>
> Best regards,
> Marian
>
> Ricardo Martinez wrote:
> > Hello.
> > When i start the STUN server from iptel (mystun) with this
> command line :
> > % ./server.exe -D
> > as a deamon.
> > How can i check which is my bind address and my alternative
> bind address?
> >
> > Thanks!
> >
> > Ricardo.-
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
> --
> Voice System
> http://www.voice-system.ro
>
