I just read the "ser status update" from jiri and he wrote in the "new
features" section:
- 3261-alignment: support for TCP and loose-routing
So I think this will be fixed in the next release?
Klaus
> -----Original Message-----
> From: Klaus Darilion
> Sent: Wednesday, February 05, 2003 11:19 AM
> To: serusers(a)lists.iptel.org
> Cc: Aymeric Moizard
> Subject: [Serusers] Routing with ser / Problems with linphone
>
>
> Hello!
>
> I've installed ser (sip:obelix.ict.tuwien.ac.at) und want'ed
> to use it with linphone, so I configured linphone to use ser
> as outbound proxy and registrar. The registration works fine
> but not the INVITEs. ser answers to an INVITE from linphone
> with a 404 Not Found and I found out that the problem is the
> "Route" header in the INVITE from linphone.
>
> Route: <sip:obelix.ict.tuwien.ac.at;lr>
>
> When I remove the header from the invite and send it manually
> (sipsak) the ser proxy accepts the invite and forwards it. I
> think ser reacts wrong because if an RFC3261 proxy gets an
> request with a route header which points to itself it should
> remove the header and forward the request.
>
> Than I tried the same with an different proxy in the route header:
>
> Route: <sip:iptel.org;lr>
>
> Now, my ser proxy (sip:obelix.ict.tuwien.ac.at) accepts the
> request and forwards it to iptel.org, but it rewrites the
> invite to: INVITE sip:iptel.org;lr SIP/2.0
>
> Thats what an RFC2543 proxy would do, but not an RFC3261
> proxy, which only is allowed to do that if the route header
> has no "lr" parameter or if he is responsible for the domain
> in the request URI. If there is an "lr" parameter the proxy
> must not change the request URI.
>
> If I'm wrong please let me know.
>
> I use ser 0.8.10 with the standard config file and linphone
> 0.9.1 & 0.10.0
>
> Regards,
> Klaus
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hello!
I've installed ser (sip:obelix.ict.tuwien.ac.at) und want'ed to use it
with linphone, so I configured linphone to use ser as outbound proxy and
registrar. The registration works fine but not the INVITEs. ser answers
to an INVITE from linphone with a 404 Not Found and I found out that the
problem is the "Route" header in the INVITE from linphone.
Route: <sip:obelix.ict.tuwien.ac.at;lr>
When I remove the header from the invite and send it manually (sipsak)
the ser proxy accepts the invite and forwards it. I think ser reacts
wrong because if an RFC3261 proxy gets an request with a route header
which points to itself it should remove the header and forward the
request.
Than I tried the same with an different proxy in the route header:
Route: <sip:iptel.org;lr>
Now, my ser proxy (sip:obelix.ict.tuwien.ac.at) accepts the request and
forwards it to iptel.org, but it rewrites the invite to:
INVITE sip:iptel.org;lr SIP/2.0
Thats what an RFC2543 proxy would do, but not an RFC3261 proxy, which
only is allowed to do that if the route header has no "lr" parameter or
if he is responsible for the domain in the request URI. If there is an
"lr" parameter the proxy must not change the request URI.
If I'm wrong please let me know.
I use ser 0.8.10 with the standard config file and linphone 0.9.1 &
0.10.0
Regards,
Klaus
Nils,
Thanks! Setting listen=sip.voiping.com worked. I had listen=192.70.239.1
(which the reverse DNS records resolve to: voiping.com). So, that worked.
Thanks for the help and the quick response.
Regarding "not forking" it was purely for debugging purposes. It'll work
now with fork=yes.
Now I have:
fork=yes
log_stderr=no
alias=voiping.comlisten=sip.voiping.com
Later,
L.
-----Original Message-----
From: Nils Ohlmeier [mailto:nils@ohlmeier.de]
Sent: Tuesday, February 04, 2003 1:57 PM
To: Lenny Tropiano; serusers(a)lists.iptel.org
Cc: 'VoIPing, LLC (IT Consulting)'
Subject: Re: [Serusers] Initial setup on FreeBSD of ser using Cisco 7960 SIP
Hi,
On Tuesday 04 February 2003 19:09, Lenny Tropiano wrote:
> I'm "playing" with SER trying to understand how it all works, and to
> be completely honest I haven't finished reading all the docs.
> Basically I'm using the default ser.cfg (installed in
> /usr/local/etc/ser) and I have two 7960s with the latest SIP code. I
> know it works, since I can call the other phone with our
> user(a)iptel.org extension ... The 2nd line is setup to register to
> sip.voiping.com [192.70.239.1]. Immediately the phone gets to trying,
> I have ser running to "not fork, debug 3, and log to stderr".
Is their any special reason why you not allow to fork?
Because from the repsonse packet bellow you can see that your request is
going
through end (nearly) endless loop. The via_cnt==12 means the request was
forwarded 12 times (before it was stoped by the maxfws rule).
I guess that ser cant detect that sip.voiping.com is one of its "myself"
names. Maybe this is caused by the 'fork=no'. If not you can try to add
'listen=sip.voiping.com' to your configuration or the commandline switch '-l
sip.voiping.com'. This should fix your problem.
Regards
Nils Ohlmeier
I'm "playing" with SER trying to understand how it all works, and to be
completely honest I haven't finished reading all the docs. Basically I'm
using the default ser.cfg (installed in /usr/local/etc/ser) and I have two
7960s with the latest SIP code. I know it works, since I can call the other
phone with our user(a)iptel.org extension ... The 2nd line is setup to
register to sip.voiping.com [192.70.239.1]. Immediately the phone gets to
trying, I have ser running to "not fork, debug 3, and log to stderr".
ngrep indicates the following:
U 192.70.239.150:53087 -> 192.70.239.1:5060
REGISTER sip:sip.voiping.com SIP/2.0..Via: SIP/2.0/UDP
192.70.239.150:5060..From: sip:lenny@sip.voiping.com..To: sip:lenny@sip.
voiping.com..Call-ID:
00036b54-b62d1fe8-314ef9d4-1cca7f7b@192.70.239.150..Date: Tue, 04 Feb 2003
18:07:32 GMT..CSeq: 101 REGIST
ER..User-Agent: CSCO/4..Contact:
<sip:lenny@192.70.239.150:5060>..Content-Length: 0..Expires: 3600....
#
U 192.70.239.1:5060 -> 192.70.239.150:5060
SIP/2.0 483 Too Many Hops..Via: SIP/2.0/UDP 192.70.239.150:5060..From:
sip:lenny@sip.voiping.com..To: sip:lenny@sip.voiping.com
;tag=e755c894a31056969aa313d5267cd575.f7b7..Call-ID:
00036b54-b62d1fe8-314ef9d4-1cca7f7b@192.70.239.150..CSeq: 101 REGISTER..Se
rver: Sip EXpress router (0.8.10 (i386/freebsd))..Content-Length:
0..Warning: 392 192.70.239.1:5060 "Noisy feedback tells: pid=
71976 req_src_ip=192.70.239.1 in_uri=sip:sip.voiping.comout_uri=sip:sip.voiping.com via_cnt==12"....
I would appreciate any advice in getting this going (btw, I compiled the
version of SER 0.8.10 and didn't use the already compiled binaries).
Thanks,
Lenny
---
Lenny Tropiano E-mail: lenny(a)voiping.com
Partner, Networking Specialist Pager: pager-lenny(a)voiping.com
VoIPing, LLC URL: http://www.voiping.com/
PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647]
Folks,
We are currently trying to implement owerflow routing with sip and b2bua.
Our network setup looks like the following (two GWs here is for simplicity,
actually there would be dozens of them):
---------
/--|PSTN GW|-\
---- ----------------- ------- /~~~~~~\/ --------- \ /~~~~~~~~\
|UA|--|PROXY/REGISTRAR|--|B2BUA|--<IP CLOUD> <PSTN CLOUD>
---- ----------------- ------- \______/\ --------- / \________/
\--|PSTN GW|-/
---------
Since potentially each destination in the PSTN could be reached through
more than one GW we would like to use that for adding some more robustness
to the system, beause from time to time some of gateways might be unavailable
for one reason of another (network outage, maintenance, overload etc.).
t_on_negative() looks like a pretty suitable feature for the job modulo
that we need to add some scheme for distinguishing real failures, such as
"number is busy", from transient ones.
The problem here is that b2bua is unable to do prefix-based routing, while
we can't put b2bua between the UA and PROXY because for accounting reasons
we should be able to get from b2bua IP number of the gateway the call was
forwarded to. Therefore, we do gateway selection based on prefix in ser
(using rewritehostport) and then just forward request to the b2bua using
t_relay_to(). To catch failures and perform retries we use t_on_negative()
and number of reply_route[] blocks and it is where the problem lies -
after appending a new branch ser forwards the request to the host:port
specified in the uri directly, but not through the b2bua.
Attached patch adds a new variable sticky_relay_to, which if set to non-zero
value instructs ser to record proxy address to which transaction was
originally forwarded with t_relay_to(). On failure ser forwards request to
that address if another branch was appended in reply_route[].
I think that it is generally useful feature and it would be nice to see
it integrated into the next release.
Thanks!
-Maxim
Hi,
I have installed SER on linux and created a few users. I can register
these users from SIP phones and soft-phones and can also establish
connections between them. I have also created a user on the
iptel.org server.
I can ring from a client registered with my domain (cs.stir.ac.uk) the
client registered with iptel.org. It works perfectly. However, ringing
from the client registered with iptel.org a client registered locally
doesn't work.
Is this a config problem? How can I fix this?
Any help is greatly appreciated!
Best regards,
Mario Kolberg
--
Mario Kolberg phone: +44 (0)1786 46 7440
Lecturer in Computing Science fax : +44 (0)1786 46 4551
email: mko(a)cs.stir.ac.uk
Department of Computing Science and Mathematics
University of Stirling
Stirling FK9 4LA
Scotland, UK
--
The University of Stirling is a university established in Scotland by
charter at Stirling, FK9 4LA. Privileged/Confidential Information may
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and any action taken or omitted to be taken in reliance on it, is
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immediately if you or your employer do not consent to Internet email
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business of the University of Stirling shall be understood as neither
given nor endorsed by it.
Hello,
I'm trying to use the presence agent that is in the CVS version.
I added the module and the lines
----
if (method=="SUBSCRIBE") {
subscribe("registrar");
break;
};
----
to my ser.cfg file. By changing this I get the following error message when
trying to "sign in" with the msn messenger UA:
----
15(23129) ERROR: t_reply: cannot send a t_reply to a message for which no
T-state has been established
15(23129) send_reply(): Error while sending 200 OK
----
What does this mean?
I admit I don't know a lot about the details of the SIP protocol, but if you'd
give me a hint on where to search, it'd help me a lot.
Is the presence agent in CVS already usable at all? If yes, could someone
please provide me a working config file?
Thanks in advance,
Stefan Schmidt
Hi,
I have set up the SER and its working. The only problem is I cant get the routing logic to simultaneously route to either the User Loc or the PSTN. Its only doing one of the other.
I am new to SIP, can u show me a sample script that will do both at the same time?
Best regards,
Phillip
Hi,
I have successfully (I think that) installed SER server with ser-mysql
package.
I have set the environment variable 'SIP_DOMAIN="192.168.11.237"' (that is
server IP address).
I have succesfully added a user to subscriber table: 'serctl add ivan
ivanpswd ivan(a)hotmail.com', but registering WM to server fails:
URI = 'ivan(a)192.168.11.237'
Passwowd = 'ivanpswd'
User = 'ivan'
This is my ser.cfg file:
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
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"
# ------------------ module loading ----------------------------------
loadmodule "//usr/lib/ser/modules/mysql.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"
loadmodule "//usr/lib/ser/modules/auth.so"
# ----------------- setting module-specific parameters ---------------
modparam("usrloc", "db_mode", 2)
modparam("auth", "db_url", "sql://ser:heslo@localhost/ser")
modparam("auth", "calculate_ha1", yes)
modparam("auth", "password_column", "password")
# ------------------------- request routing logic -------------------
alias="engiweb.com"
alias="sip.engiweb.com"
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Message too big");
break;
};
rewriteFromRoute();
if (uri=~"^sip:(.+@)?(192\.168\.11\.237|(sip\.)?engiweb\.com)([:;\?].*)?$")
{
if (method=="REGISTER") {
if (!www_authorize("192.168.11.237", "subscriber")) {
if (!proxy_authorize("192.168.11.237", "subscriber")) {
www_challenge("engiweb.org", "0");
break;
};
save("location");
break;
};
lookup("aliases");
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
What could be the problem?
Best regards.
Ivan Vignola.
_________________________________________________________________
MSN Extra Storage: piena libertà di esprimersi e comunicare
http://www.msn.it/msnservizi/es/extra_storage_tag/
Hi Daniel,
The problem related with the permition to use de jabber gateway seems to be
solved.
Now I have a new problem. In the same scenario (Messenger sending a message
to Jabber client) the messenger receives from the jabber gw the following
error message:
" ERROR:Connection to Jabber server lost. You have to login to Jabber server
again (join the conferences again that you were participating,
too)<?xml:namespace prefix = o ns =
"urn:schemas-microsoft-com:office:office" />"
Can you figure what is happening here?
Thanks,
Toni
-----Original Message-----
From: Daniel-Constantin MIERLA [mailto:mierla@fokus.fraunhofer.de]
Sent: sexta-feira, 31 de Janeiro de 2003 12:51
To: Toni Barata (EPS)
Cc: 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] user permission problem on ser-jabber gw
Hello,
the SIP_ID field must be the SIP URI of the SIP user (e.g. '
sip:user1@mydomain.com <mailto:sip:user1@mydomain.com> ') -- that is for
fast matching - From tag includes sip: in address of the origin. If you
check the manual or the mail I pointed, the samples show that.
Toni Barata (EPS) wrote:
Hi Daniel,
I have done this (creating the database sip_jab) before sending my first
email.
Regarding the association of SIP users with Jabber IDs, I have added an
entry on table jusers this way on mysql:
mysql> INSERT INTO jusers VALUES
('1','user1','user1',' user1(a)mydomain.com <mailto:user1@mydomain.com>
','0');
Should it be done this way or there are others ways to do it?
If you take the sources of the module, in directory 'doc' you can find some
perl or php scripts which can help you. You have to change the values from
the beginning of scripts according with your configuration.
The same problem persists.
Try again now.
Regards,
Daniel
Regards,
Toni
-----Original Message-----
From: Daniel-Constantin MIERLA [ mailto:mierla@fokus.fraunhofer.de
<mailto:mierla@fokus.fraunhofer.de> ]
Sent: quinta-feira, 30 de Janeiro de 2003 19:54
To: Toni Barata (EPS)
Cc: ' serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> '
Subject: Re: [Serusers] user permission problem on ser-jabber gw
Hello,
you have to create the database for SIMPLE2Jabber gateway and after that
you have to associate SIP users with Jabber IDs.
I wrote a manual for gateway, but is more related to next ser release.
Anyway, the database is the same, exported methods too, only some
parameters has changed. The database can be created using sql script
from CVS
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/jabber/doc/
<http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/modules/jabber/doc
/>
xjab.sql?rev=1.1&content-type=text/vnd.viewcvs-markup
or from manual
http://www.iptel.org/ser/doc/jabgw/xjab-manual.html#5._Admins_guide
<http://www.iptel.org/ser/doc/jabgw/xjab-manual.html#5._Admins_guide>
It might provide a good help to read next message (especially last part
of it) posted on this list
http://lists.iptel.org/pipermail/serusers/2002-December/000049.html
<http://lists.iptel.org/pipermail/serusers/2002-December/000049.html>
After that if you still have problems do not hesitate to post a new
message on the list.
Best regards,
Daniel
Toni Barata (EPS) wrote:
Hi folks,
( I think!) I have successfully installed SER server with the ser-mysql and
ser-jabber packages ( I have also a mysql server and a Jabber server
running).
For testing I have established a SIP IM session between two MS Messengers
and it did work.
After this I have tried to establish a IM session between a user SIP (MS
Messenger) and a user Jabber (Exodus), but when the messenger sends a
message for the jabber gateway the following error occurs:
"ERROR: Your message was not sent. You do not have permision to use the
gateway."
What could be the problem here?
Best regards,
Toni
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