In debugging yet another SIP hardware device I
found that it was not authenticating after an
INVITE. That is, the device sends me 'INVITE', I say
'401', it says 'ACK', but I never hear from it again.
I am expecting another INVITE with the proper credentials...
In talking with the hardware vendor they let me know that
I should be sending back a 407 instead of a 401.
I have always done a www_challenge sequence, but,
I wonder if that is truly proper. After all, SER is
a proxy. Should I be challenging with a 'proxy_challenge'?
Certainly either should work, because the device you
are communicating with could be a UA or a PROXY.
What is the 'proper' thing to do?
I guess I have always thought that the 'proxy_challenge'
was for one proxy server communicating with another.
However I don't see how that can be now, because the
originating proxy server has no mechanism to provide
authentication credentials...
The more I learn, the less I know,
---greg
Please find attached the results of the "ngrep" entry.
>
> From: 'Jan Janak' <jan(a)iptel.org>
> Date: Thu, 4 Dec 2003 17:00:49 +0100
> To: Andy Vander Woude <andyv(a)sympatico.ca>
> CC: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Residential Gateways
>
> Hello,
>
> please always CC the mailing list. If you get parsing error then the
> SIP messages from the gateways are probably not formatted correctly (or
> there is a bug in ser).
>
> Please create SIP message dumps and send them to me.
>
> Jan.
>
> On 03-12 23:37, Andy Vander Woude wrote:
> > The residential gateways are "Registering" with the server however when I
> > try to make a call between the two RG's all I get is a busy signal and
> > parsing errors. The Transaction Status - Completion Status 4XXX increments
> > up in numbers as I try the connections. I have attached the ser.cfg file.
> >
> > -----Original Message-----
> > From: Jan Janak [mailto:jan@iptel.org]
> > Sent: Wednesday, December 03, 2003 11:23 AM
> > To: Andy Vander Woude
> > Cc: serusers(a)lists.iptel.org
> > Subject: Re: [Serusers] Residential Gateways
> >
> > Hello,
> >
> > I would recommend you to create SIP message dumps (using ngrep or
> > ethereal) and either look at them or send them to us. We would also need
> > to see your ser.cfg. Without this information it is hard to say what is
> > wrong.
> >
> > A quick guess: You didn't configure ser properly so it doesn't send SIP
> > INVITE messages to the gateways.
> >
> > Jan.
> >
> > On 30-11 19:59, Andy Vander Woude wrote:
> > >
> > > Running Redhat V9 and have installed the SIP Express Router (ser)
> > > packages. Have two Allied Telesyn Residential Gateways (192.168.20.2 and
> > > 192.168.20.3) and laptop (192.168.20.4) running ser connecting to a
> > > managed switch IP address (192.168.20.1 (gateway))
> > > The RG's are configured to access the Proxy server, & Domain server at
> > > 192.168.20.4 and the Gateway as 192.168.20.1 The Domain is called
> > > ati.com.
> > >
> > > When connected all together I get a dial tone on the phones, however
> > > when I dial the # configured on the RG's I do not get a ring and they do
> > > not register with the ser server. When I type in the "ser start" command
> > > I see the alias entries as "127.0.0.1 localhost localdomain localhost"
> > > and "192.168.20.4 localhost.ati.com localhost"
> > >
> > > The entries in the ser.cfg relating to the localdomain have been changed
> > > to ati.com
> > > I have added the 192.168.20.4 to the hosts.cfg relating to the ati.com
> > > domain.
> > >
> > > Any idea as to why this configuration is not working. This is a lab
> > > situation with no outside connections.
> > > Thank you.
> >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
>
Howdy!
I'm having some trouble with the nathelper module and certain types of
broadband routers (ie d-link 604 & d-link 624). I'll try to explain the
situation below and hope that someone is willing to help me out, because
i'm stuck.
In short, the setup is a ata-186 box (which is symmetric) behind a
d-link 604 (which isn't symmetric at all times).
The nathelper module included in the distribution (both 0.8.11 and
0.8.12) has a function called fix_nated_contact(). fix_nated_contact()
rewrites the contact-header with the source-ip & source-port of the packet.
However, in some cases (ie non-symmetric d-link router between the
ata-box and the internet) this is a problem since the d-link router
sometimes rewrites the source-port which is then used as a location in
ser. When the session has timed out on the d-link (doesn't really seem
to help with the natping) the location-information in ser is no longer
valid.
Is there any reason why the nathelper rewrites the port in the
contact-header? If the client is symmetric the source-port and the port
in the contact-header shouldn't differ anyway? I trust there is a
reason, i just dont see it ;)
On a side-note, when glancing at nathelper.c it looks as if the int len
is calculated with the original values of the header, then filled
through snprintf with values which are not 100% positively the same
length (msg->rcv.src_port). Isn't it for example possible that the port
in the header is 5060 but the source_port is 22444 (which is one
character longer than the length of len is calculated to).
I hope someone can shed some light over the matter.
/Martin
Hmm.. I'm having a hard time getting it to work.. Here is the configuration
I've got on my ser box.
#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd)
fork=yes
#log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
alias=acme.dom
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/ser/modules/nathelper.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
#inserted by klaus
if (method=="INVITE") {
record_route();
fix_nated_contact();
force_rtp_proxy();
t_on_reply("1");
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
#inserted by klaus
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
if (status=~"[0-9][0-9]")
fix_nated_contact();
force_rtp_proxy();
}
-----Original Message-----
From: TeleSIP [mailto:ricvil@telesip.net]
Sent: Wednesday, December 03, 2003 4:15 PM
To: Darren Sessions
Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
No...All you need to do is to compile it and the run it.
Regards,
Andres
----- Original Message -----
From: "Darren Sessions" <dsessions(a)ionosphere.net>
To: "TeleSIP" <ricvil(a)telesip.net>
Cc: <serusers(a)lists.iptel.org>
Sent: Wednesday, December 03, 2003 3:39 PM
Subject: RE: [Serusers] NatHelper & Portaone RTP Proxy
>
> Do I need to do anything special with rtpproxy?
>
> -----Original Message-----
> From: TeleSIP [mailto:ricvil@telesip.net]
> Sent: Wednesday, December 03, 2003 12:44 PM
> To: Darren Sessions
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
>
>
> Take a loot at the nathelper_rtp.cfg file under the modules/nathelper
> directory. It contains the basics of what you need.
>
> The funcionality is basically determined by these lines:
>
> if (method=="INVITE") {
> record_route();
> if (isflagset(1)) { # ATA ?
> force_rtp_proxy();
> };
> /* set up reply processing */
> t_on_reply("1");
> };
> .
> .
> .
> onreply_route[1] {
> if (status=~"1[0-9][0-9]" && search("Server: Cisco ATA.*"))
> fix_nated_contact();
> force_rtp_proxy();
> }
>
>
> ----- Original Message -----
> From: "Darren Sessions" <dsessions(a)ionosphere.net>
> To: "TeleSIP" <ricvil(a)telesip.net>
> Cc: <serusers(a)lists.iptel.org>
> Sent: Wednesday, December 03, 2003 12:20 PM
> Subject: RE: [Serusers] NatHelper & Portaone RTP Proxy
>
>
> > Hmmm.. You wouldn't by chance have a "scrubbed" version of your
> > configuration file I could look at would you? It would be most helpful.
> >
> > Thanks for everything,
> >
> > - Darren
> >
> > -----Original Message-----
> > From: TeleSIP [mailto:ricvil@telesip.net]
> > Sent: Wednesday, December 03, 2003 12:18 PM
> > To: Darren Sessions
> > Cc: serusers(a)lists.iptel.org
> > Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
> >
> >
> > We have nathelper/rtpproxy/SER all on the same server. The ser.cfg must
> be
> > configured to invoke the nathelper/rtproxy functions as needed.
> >
> > I cannot comment on SEMS since we don't use it.
> >
> > ----- Original Message -----
> > From: "Darren Sessions" <dsessions(a)ionosphere.net>
> > To: "TeleSIP" <ricvil(a)telesip.net>
> > Cc: <serusers(a)lists.iptel.org>
> > Sent: Wednesday, December 03, 2003 12:11 PM
> > Subject: RE: [Serusers] NatHelper & Portaone RTP Proxy
> >
> >
> > > My next questions is then, what configuration changes do I make to the
> > > nathelper server to make it forward requests made from the endpoints
to
> > the
> > > primary ser box? and how does voicemail come into play (using sems)?
> Does
> > > the nathelper server "know" about all the servers or simply does the
> > > signaling traffic and rtp traffic get routed through the nathelper
box?
> > >
> > > Thanks a ton for all the help!
> > >
> > > - Darren
> > >
> > > -----Original Message-----
> > > From: TeleSIP [mailto:ricvil@telesip.net]
> > > Sent: Wednesday, December 03, 2003 10:46 AM
> > > To: Darren Sessions
> > > Cc: serusers(a)lists.iptel.org
> > > Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
> > >
> > >
> > > It works great on RedHat. We have it running now for about 4 months
on
> > our
> > > production servers and it has never crashed
> > >
> > > ----- Original Message -----
> > > From: "Darren Sessions" <dsessions(a)ionosphere.net>
> > > To: "TeleSIP" <ricvil(a)telesip.net>
> > > Cc: <serusers(a)lists.iptel.org>
> > > Sent: Wednesday, December 03, 2003 10:16 AM
> > > Subject: RE: [Serusers] NatHelper & Portaone RTP Proxy
> > >
> > >
> > > > I'm starting over on a Redhat Linux box right now.
> > > >
> > > > I'll let you know how it goes..
> > > >
> > > >
> > > > -----Original Message-----
> > > > From: TeleSIP [mailto:ricvil@telesip.net]
> > > > Sent: Wednesday, December 03, 2003 10:15 AM
> > > > To: Darren Sessions
> > > > Cc: serusers(a)lists.iptel.org
> > > > Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
> > > >
> > > >
> > > > Darren,
> > > >
> > > > 3 months ago I tried to get this to work on Solaris 8. Maxim also
put
> > > some
> > > > time into it but I was never able to get it to compile. If you have
> > > better
> > > > luck please let us all know.
> > > >
> > > > Thanks,
> > > > Andres.
> > > >
> > > > ----- Original Message -----
> > > > From: "Darren Sessions" <dsessions(a)ionosphere.net>
> > > > To: "Andrei Pelinescu-Onciul" <pelinescu-onciul(a)fokus.fraunhofer.de>
> > > > Cc: <serusers(a)lists.iptel.org>
> > > > Sent: Wednesday, December 03, 2003 9:40 AM
> > > > Subject: RE: [Serusers] NatHelper & Portaone RTP Proxy
> > > >
> > > >
> > > > > I am having problems compiling rtpproxy on a Sun Netra 1400t with
> > > Solaris
> > > > 9.
> > > > >
> > > > > root:/export/home/rtpproxy # make all
> > > > > cc -o main.o -c main.c
> > > > > In file included from main.c:45:
> > > > > myqueue.h:40:23: sys/cdefs.h: No such file or directory
> > > > > main.c:51:17: err.h: No such file or directory
> > > > > main.c: In function `main':
> > > > > main.c:486: structure has no member named `sun_len'
> > > > > make: *** [main.o] Error 1
> > > > >
> > > > > I am new to this stuff, so everyone will have to bear with me.
> > > > >
> > > > > I can't seem to find cdefs.h or err.h. I did however throw the
> > > > Makefile.gnu
> > > > > up as Makefile and uncommented the section labeled "for Solaris".
> > > > >
> > > > > Any help would be appreciated! :)
> > > > >
> > > > > Thanks,
> > > > >
> > > > > - Darren
> > > > >
> > > > > -----Original Message-----
> > > > > From: Andrei Pelinescu-Onciul
> > > > > [mailto:pelinescu-onciul@fokus.fraunhofer.de]
> > > > > Sent: Tuesday, December 02, 2003 5:36 PM
> > > > > To: Darren Sessions
> > > > > Cc: serusers(a)lists.iptel.org
> > > > > Subject: Re: [Serusers] NatHelper & Portaone RTP Proxy
> > > > >
> > > > >
> > > > > On Dec 02, 2003 at 17:10, Darren Sessions
<dsessions(a)ionosphere.net>
> > > > wrote:
> > > > > > Someone correct me if I'm wrong, but is the typical setup for
> > > Nathelper
> > > > > and
> > > > > > Portaone RTP proxy work on seperate servers with dual network
> cards?
> > > > >
> > > > > No. nathelper use unix sockets to communicate with rtprpoxy so
they
> > must
> > > > > be located on the same box (ser runing nathelper and rtproxy).
> > > > >
> > > > > Also it is better to run/configure them on only one interface (to
> > avoid
> > > > > packets with different source address than the destination of the
> > > intial
> > > > > packet going back to the nat).
> > > > >
> > > > > >then
> > > > > > talking to another SER box doing registrations and call routing?
> and
> > > the
> > > > > > Portaone RTP taking the voice packets?
> > > > >
> > > > >
> > > > > Andrei
> > > > >
> > > > > _______________________________________________
> > > > > Serusers mailing list
> > > > > serusers(a)lists.iptel.org
> > > > > http://lists.iptel.org/mailman/listinfo/serusers
> > > > >
> > > > >
> > > >
> >
Hello,
please always CC the mailing list. If you get parsing error then the
SIP messages from the gateways are probably not formatted correctly (or
there is a bug in ser).
Please create SIP message dumps and send them to me.
Jan.
On 03-12 23:37, Andy Vander Woude wrote:
> The residential gateways are "Registering" with the server however when I
> try to make a call between the two RG's all I get is a busy signal and
> parsing errors. The Transaction Status - Completion Status 4XXX increments
> up in numbers as I try the connections. I have attached the ser.cfg file.
>
> -----Original Message-----
> From: Jan Janak [mailto:jan@iptel.org]
> Sent: Wednesday, December 03, 2003 11:23 AM
> To: Andy Vander Woude
> Cc: serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Residential Gateways
>
> Hello,
>
> I would recommend you to create SIP message dumps (using ngrep or
> ethereal) and either look at them or send them to us. We would also need
> to see your ser.cfg. Without this information it is hard to say what is
> wrong.
>
> A quick guess: You didn't configure ser properly so it doesn't send SIP
> INVITE messages to the gateways.
>
> Jan.
>
> On 30-11 19:59, Andy Vander Woude wrote:
> >
> > Running Redhat V9 and have installed the SIP Express Router (ser)
> > packages. Have two Allied Telesyn Residential Gateways (192.168.20.2 and
> > 192.168.20.3) and laptop (192.168.20.4) running ser connecting to a
> > managed switch IP address (192.168.20.1 (gateway))
> > The RG's are configured to access the Proxy server, & Domain server at
> > 192.168.20.4 and the Gateway as 192.168.20.1 The Domain is called
> > ati.com.
> >
> > When connected all together I get a dial tone on the phones, however
> > when I dial the # configured on the RG's I do not get a ring and they do
> > not register with the ser server. When I type in the "ser start" command
> > I see the alias entries as "127.0.0.1 localhost localdomain localhost"
> > and "192.168.20.4 localhost.ati.com localhost"
> >
> > The entries in the ser.cfg relating to the localdomain have been changed
> > to ati.com
> > I have added the 192.168.20.4 to the hosts.cfg relating to the ati.com
> > domain.
> >
> > Any idea as to why this configuration is not working. This is a lab
> > situation with no outside connections.
> > Thank you.
>
> > _______________________________________________
> > Serusers mailing list
> > serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
In both directions or only in one direction?
You can use ethereal to capture the RTP streams and save them to disc
(can help you to find out if the problem is at the sender or at the
receiver). Take a look at the RTP headers and the used codec. Try to
disable some codec or change the preferrences of the codec.
regards,
Klaus
> -----Original Message-----
> From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
> Sent: Thursday, December 04, 2003 3:58 PM
> To: serusers
> Subject: [Serusers] xten and Mitel 5055
>
>
> Hi,
>
> apologies for using this email list for something which is
> not directly
> linked to ser.
>
> I'm having a weird problem when setting up a voice connection
> between an
> Xten X-Lite UA and a Mitel 5055 SIP phone (connected via a ser proxy
> server). The session initiation is all fine, but the media session is
> not - the voice transmitted sounds a bit like a robot and is
> impossible
> to understand. I assumed this is a codec problem but strangly this
> problem ONLY occurs between the Xten UA and the Mitel phone.
> It does not
> occur between Xten and a Pingtel phone nor between Mitel 5055 and
> Pingtel, nor between Ubiquity UA and Mitel 5055. As far as I know all
> use the G711 codec.
>
> Has anybody experienced this problem also?
>
> Thank you for your help.
> Mario Kolberg
>
>
>
> --
> The University of Stirling is a university established in Scotland by
> charter at Stirling, FK9 4LA. Privileged/Confidential Information may
> be contained in this message. If you are not the addressee indicated
> in this message (or responsible for delivery of the message to such
> person), you may not disclose, copy or deliver this message to anyone
> and any action taken or omitted to be taken in reliance on it, is
> prohibited and may be unlawful. In such case, you should destroy this
> message and kindly notify the sender by reply email. Please advise
> immediately if you or your employer do not consent to Internet email
> for messages of this kind.
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hi,
please keep list in CC to enable other users to learn from the results.
On Thursday 04 December 2003 15:20, Franz Edler wrote:
> > From: Nils Ohlmeier Sent: Thursday, December 04, 2003 2:32 PM
> >
> > Ohh, sorry. Seems i read not carefully enough.
> > But what was the output of '/usr/sbin/ser_mysql.sh create'? If there is
> > no Ser database after this command, i assume that there was some kind of
> > error message as result of the command.
>
> This is the dialog, that happens at '/usr/sbin/ser_mysql.sh create´':
>
> MySql password for root: <my Linux root password>
> Doamin (realm) for the default user 'admin': < value of variable
> SIP_DOMAIN>
>
> /usr/sbin/ser_mysql.sh: line 1: gen_ha1: command not found
> HA1 calculation failed.
This is very strange because the gen_ha1 is included in the
ser-0.8.11-0.i386.rpm as /usr/sbin/gen_ha1.
Did you installed the basic ser rpm?
It is a dependency for the ser-mysql rpm, so i guess yes. In that case my only
assumption is that for some very strange reason /usr/sbin/ is not included in
your PATH.
Regards
Nils
Hi,
apologies for using this email list for something which is not directly
linked to ser.
I'm having a weird problem when setting up a voice connection between an
Xten X-Lite UA and a Mitel 5055 SIP phone (connected via a ser proxy
server). The session initiation is all fine, but the media session is
not - the voice transmitted sounds a bit like a robot and is impossible
to understand. I assumed this is a codec problem but strangly this
problem ONLY occurs between the Xten UA and the Mitel phone. It does not
occur between Xten and a Pingtel phone nor between Mitel 5055 and
Pingtel, nor between Ubiquity UA and Mitel 5055. As far as I know all
use the G711 codec.
Has anybody experienced this problem also?
Thank you for your help.
Mario Kolberg
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may
be contained in this message. If you are not the addressee indicated
in this message (or responsible for delivery of the message to such
person), you may not disclose, copy or deliver this message to anyone
and any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful. In such case, you should destroy this
message and kindly notify the sender by reply email. Please advise
immediately if you or your employer do not consent to Internet email
for messages of this kind.
I can't say it's a messenger issue. Messenger wants a contact list, and to
show you who is online and offline. Your phone doesn't do that. The phone
talks to SER to see if the person is signed in. Or does kphone and sipc
show users signed in?
Scott Morris
Enterprise Network Engineer
DOE - ORAU / ORISE
865-576-4672
-----Original Message-----
From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
Sent: Thursday, December 04, 2003 5:47 AM
To: Morris, Scott; serusers(a)lists.iptel.org
Subject: Re:[Serusers] Windows Messenger 5.0
I have made exactly the same observation. I treat it as a problem with
Messenger as it works fine between two kphone UAs or between sipc and
kphone. I'm running ser 0.8.12.
Mario
> I have 0.8.1 running on Suse 9.0. It works great with my Zultys
> phone, Polycom IP 600, and the Helmsen agent. When I try to use
> Windows Messenger 5.0 I have a slight problem.
>
> UserA signs in. UserB signs in, and sees that UserA is online, and
> there changes the UserA in the contact list on ONLINE. But UserB
> never shows as ONLINE on UserA's desktop. So UserA can't call UserB.
> The update never goes to UserA's desktop. What am I missing here?
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may be
contained in this message. If you are not the addressee indicated in this
message (or responsible for delivery of the message to such person), you may
not disclose, copy or deliver this message to anyone and any action taken or
omitted to be taken in reliance on it, is prohibited and may be unlawful.
In such case, you should destroy this message and kindly notify the sender
by reply email. Please advise immediately if you or your employer do not
consent to Internet email for messages of this kind.
Windows Messenger does not subscribe at ser. ser only forwards the
subscribe request to the requested client. Kphone uses the same
technique.
Anyway, I suggest to uninstall Windows Messenger 5.0 and install 4.6 or
4.7. For further investigation: start a packet sniffer (eg.g ngrep9 and
analyze the SIP messages (and/or post it here)
regards,
Klaus
> -----Original Message-----
> From: Morris, Scott [mailto:MorrisS@orau.gov]
> Sent: Thursday, December 04, 2003 2:29 PM
> To: 'Mario Kolberg'; serusers(a)lists.iptel.org
> Subject: RE: [Serusers] Windows Messenger 5.0
>
>
> I can't say it's a messenger issue. Messenger wants a
> contact list, and to
> show you who is online and offline. Your phone doesn't do
> that. The phone
> talks to SER to see if the person is signed in. Or does
> kphone and sipc
> show users signed in?
>
> Scott Morris
> Enterprise Network Engineer
> DOE - ORAU / ORISE
> 865-576-4672
>
>
> -----Original Message-----
> From: Mario Kolberg [mailto:mko@cs.stir.ac.uk]
> Sent: Thursday, December 04, 2003 5:47 AM
> To: Morris, Scott; serusers(a)lists.iptel.org
> Subject: Re:[Serusers] Windows Messenger 5.0
>
>
> I have made exactly the same observation. I treat it as a
> problem with
> Messenger as it works fine between two kphone UAs or between sipc and
> kphone. I'm running ser 0.8.12.
>
> Mario
>
>
> > I have 0.8.1 running on Suse 9.0. It works great with my Zultys
> > phone, Polycom IP 600, and the Helmsen agent. When I try to use
> > Windows Messenger 5.0 I have a slight problem.
> >
> > UserA signs in. UserB signs in, and sees that UserA is online, and
> > there changes the UserA in the contact list on ONLINE. But UserB
> > never shows as ONLINE on UserA's desktop. So UserA can't
> call UserB.
> > The update never goes to UserA's desktop. What am I missing here?
>
>
>
> --
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>
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