Sorry, auth was excluded from compilation by default.
Either update sip_router/Makefile from the repository
or edit sip_router/Makefile, look for line containing excluded_modules,
remove auth from it and do again make all.
Jan.
PS: Please CC the mailing list too.
On 12-03 16:39, David Beckemeyer wrote:
>
> Thanks for your help.
>
> For some reason modules/auth/auth.so didn't get built so that ser.cfg
> doesn't start.
>
> ser: ERROR: load_module: could not open …
[View More]module <./modules/auth/auth.so>: ./modules/auth/auth.so: cannot open shared object file: No such file or directory
> ser: parse error (40,13-36): failed to load module
> ser: parse error (92,26-27): unknown command, missing loadmodule?
> ser: INFO: signal 15 received
>
> It looks like the Makefile nevver tried to build that module.
>
> Other modules got built, as follows:
> modules/acc/acc.so
> modules/auth_db/auth_db.so
> modules/dbtext/dbtext.so
> modules/domain/domain.so
> modules/enum/enum.so
> modules/exec/exec.so
> modules/group/group.so
> modules/maxfwd/maxfwd.so
> modules/pike/pike.so
> modules/print/print.so
> modules/registrar/registrar.so
> modules/rr/rr.so
> modules/sl/sl.so
> modules/textops/textops.so
> modules/tm/tm.so
> modules/uri/uri.so
> modules/usrloc/usrloc.so
> modules/vm/vm_mod.so
>
> On Thu, Mar 13, 2003 at 01:15:26AM +0100, Jan Janak wrote:
> > Hello David,
> >
> > attached please find updated ser.cfg with authentication enabled using
> > plaintext files instead of mysql.
> >
> > Do the following:
> > 1) Get the latest CVS snapshot (yes, again)
> > 2) Compile it, but do not install it (don't do make install)
> > 3) Create a directory where your plaintext files containing database
> > tables will be stored
> > 4) Put attached "subscriber" and "location" files in the directory
> > 5) Edit attached ser.cfg
> > 6) Replace /home/janakj/ser with path to the directory (2 occurrencies)
> > 7) save ser.cfg in sip_router directory (the CVS snaphot)
> > 7) Edit subscriber and replace jan with your username, heslo with your
> > password and iptel.org with your domain
> > 8) cd sip_router
> > 9) ./ser -f ser.cfg
> >
> > Please report any problems to us.
> >
> > Jan.
> >
> > PS: See sip_router/modules/dbtext/Readme for description of format of
> > the plaintext files.
> >
> > On 12-03 13:57, David Beckemeyer wrote:
> > >
> > > Unstable indeed. I built SER from CVS and even using the
> > > stock (no DB) ser.cfg, it won't start, and reports:
> > >
> > > ERROR: bad config file (16 errors)
> > >
> > > ser: parse error (36,13-47): failed to load module
> > > ser: set_mod_param(): Parameter not found
> > > ser: parse error (46,34-35): Can't set module parameter
> > > ser: parse error (70,36-37): unknown command, missing loadmodule?
> > > ser: parse error (71,38-39): unknown command, missing loadmodule?
> > > ser: parse error (75,41-42): unknown command, missing loadmodule?
> > > ser: parse error (80,19-20): unknown command, missing loadmodule?
> > > ser: parse error (95,19-20): unknown command, missing loadmodule?
> > > ser: parse error (100,25-26): unknown command, missing loadmodule?
> > > ser: parse error (101,36-37): unknown command, missing loadmodule?
> > > ser: parse error (106,15-16): unknown command, missing loadmodule?
> > > ser: parse error (107,18-19): unknown command, missing loadmodule?
> > > ser: INFO: signal 15 received
> > >
> > > On Wed, Mar 12, 2003 at 08:40:04PM +0100, Daniel-Constantin MIERLA wrote:
> > > >
> > > >
> > > > Hello,
> > > > see dbtext for that. But notice that the module has not been thoroughly
> > > > tested. It is not included in v0.8.10 so you have to use a CVS snapshot
> > > > which is quite unstable these days due to lots of changes. If you want
> > > > to use it before the new release of ser is out you might get weird
> > > > situations, but you can provide us some help with testing ... It is your
> > > > choice ...
> > > >
> > > > -.-
> > > > Best regards,
> > > > Daniel
> > > >
> > > >
> > > > David Beckemeyer wrote:
> > > >
> > > > >I'm trying to get SER up and I need auth. I'm not a Mysql user
> > > > >and, frankly, I'd rather not need to learn it, just to get
> > > > >authentication support. I wish there were a middle-ground
> > > > >between no-auth at all and a full-blown Mysql installation.
> > > > >
> > > > >I tried the Cookbook approach, using Dan Austin's HOWTO for
> > > > >setting up Mysql for SER, but the commands do not work as shown
> > > > >so I'm left with learning Mysql through and through to debug
> > > > >why those cookbook commands don't work.
> > > > >
> > > > >I'm very familiar with SIP and I want to test SER, not a new Sql
> > > > >server DB engine. It seems like Mysql is a distraction to that
> > > > >effort.
> > > > >
> > > > >Is anyone working on something like a flat-file module to
> > > > >replace Mysql? Any pointers for the best place to start on
> > > > >developing such a module (i.e. shortcuts)?
> > > > >
> > > > >Thanks,
> > > > >
> > > > > - david
> > > > >
> > > > >_______________________________________________
> > > > >Serusers mailing list
> > > > >serusers(a)lists.iptel.org
> > > > >http://lists.iptel.org/mailman/listinfo/serusers
> > > > >
> > > > >
> > > > >
> > > >
> > > >
> > > >
> > > > _______________________________________________
> > > > Serusers mailing list
> > > > serusers(a)lists.iptel.org
> > > > http://lists.iptel.org/mailman/listinfo/serusers
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
>
> > #
> > # $Id: dbtext.cfg,v 1.2 2003/03/13 00:07:28 janakj Exp $
> > #
> > # simple quick-start config script
> > #
> >
> > # ----------- global configuration parameters ------------------------
> >
> > debug=3 # debug level (cmd line: -dddddddddd)
> > fork=yes
> > log_stderror=no # (cmd line: -E)
> >
> > /*
> > 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"
> >
> > # ------------------ module loading ----------------------------------
> >
> > # Uncomment this if you want to use SQL database
> > loadmodule "./modules/dbtext/dbtext.so"
> >
> > loadmodule "./modules/sl/sl.so"
> > loadmodule "./modules/tm/tm.so"
> > loadmodule "./modules/rr/rr.so"
> > loadmodule "./modules/maxfwd/maxfwd.so"
> > loadmodule "./modules/usrloc/usrloc.so"
> > loadmodule "./modules/registrar/registrar.so"
> >
> > # Uncomment this if you want digest authentication
> > # mysql.so must be loaded !
> > loadmodule "./modules/auth/auth.so"
> > loadmodule "./modules/auth_db/auth_db.so"
> >
> > # ----------------- setting module-specific parameters ---------------
> >
> > # -- usrloc params --
> >
> > #modparam("usrloc", "db_mode", 0)
> >
> > # Uncomment this if you want to use SQL database
> > # for persistent storage and comment the previous line
> > modparam("usrloc", "db_mode", 1)
> > modparam("usrloc", "db_url", "/home/janakj/ser")
> > modparam("auth_db", "db_url", "/home/janakj/ser")
> >
> > # -- auth params --
> > # Uncomment if you are using auth module
> > #
> > modparam("auth_db", "calculate_ha1", yes)
> > #
> > # If you set "calculate_ha1" parameter to yes (which true in this config),
> > # uncomment also the following parameter)
> > #
> > modparam("auth_db", "password_column", "password")
> >
> > # ------------------------- 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 (len_gt( max_len )) {
> > sl_send_reply("513", "Message too big");
> > break;
> > };
> >
> > # Do strict routing if pre-loaded route headers present
> > loose_route();
> >
> > # 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") {
> > if (!www_authorize("", "subscriber")) {
> > www_challenge("", "0");
> > break;
> > };
> >
> > save("location");
> > break;
> > };
> >
> > # native SIP destinations are handled using our USRLOC DB
> > if (!lookup("location")) {
> > sl_send_reply("404", "Not Found");
> > break;
> > };
> > };
> > # forward to current uri now
> > if (!t_relay()) {
> > sl_reply_error();
> > };
> >
> > }
> >
>
> > user(str) contact(str) expires(int) q(double) callid(str) cseq(int)
>
> > username(str) password(str) ha1(str) domain(str) ha1b(str)
> > jan:heslo:xxx:iptel.org:xxx
> >
>
>
>
[View Less]
I'm trying to get SER up and I need auth. I'm not a Mysql user
and, frankly, I'd rather not need to learn it, just to get
authentication support. I wish there were a middle-ground
between no-auth at all and a full-blown Mysql installation.
I tried the Cookbook approach, using Dan Austin's HOWTO for
setting up Mysql for SER, but the commands do not work as shown
so I'm left with learning Mysql through and through to debug
why those cookbook commands don't work.
I'm very familiar with SIP …
[View More]and I want to test SER, not a new Sql
server DB engine. It seems like Mysql is a distraction to that
effort.
Is anyone working on something like a flat-file module to
replace Mysql? Any pointers for the best place to start on
developing such a module (i.e. shortcuts)?
Thanks,
- david
[View Less]
Hi,
I saw many changes in CVS about radius. Is there any progress for radius?
The radius modules from release 0.8.10 are unfortunately not usable.
It seems there will be some fundamental changes in terms of radius_auth and
radius_acc. How will the new structure look?
Best regards
Yang
Hello,
We are going to update the SER Programmer's Guide (If you don't know it:
http://iptel.org/ser/devel.html ).
If you have any suggestions, ideas or even want to contribute something,
please let us know. The document will be intended mainly for 3rd party
module developers and feedback from you is very important to us.
Any feedback will be appreciated, thank you.
regards, Jan.
Sorry if you receive duplicated mail from other groups.
============================================================================
=======
Mitel Networks To Host Leading Global SIP Interoperability Event
SIPit 13 to showcase the interoperability and implementations of Session
Initiation Protocol (SIP) from over 60 companies
OTTAWA, CANADA, March 11, 2003 - Mitel Networks, a market-leader in voice,
video, and data convergence over broadband networks, today announced that it
will host and …
[View More]sponsor the 13th annual SIPit, a global SIP interoperability
testing event. The event will take place August 18 to 23 at the Brookstreet
Hotel in Ottawa, Canada and attract over 60 vendors and 150 engineers from
around the globe. SIPit is an international event endorsed by the SIP Forum
and overseen by the Internet Engineering Task Force (IETF). SIPit provides
a critical industry benchmark in the drive to make SIP the defacto standard
for real-time Internet communication services. SIP is an open,
Internet-genuine protocol for establishing and managing multi-party,
mixed-media sessions over converged networks.
The purpose of SIPit is to test the interoperability of SIP implementations
and products and perform testing on advanced features and functionality.
Vendors then have the opportunity to address any incompatibilities for their
commercial releases. The testing at SIPit also accelerates the drive
towards new drafts of the SIP specification based on new requirements and
findings at the event. SIPit is open exclusively to vendors offering
SIP-based solutions that will see market availability and is not an
exhibition, public demonstration, conference or workshop.
"The promise of SIP to enable businesses the choice and flexibility of true
open interoperability in a multi-vendor environment will gain velocity as
manufacturers collaborate in testing their standards based solutions," said
Paul Butcher, Chief Operating Officer, Mitel Networks. "The opportunity to
test our products at SIPit, with the devices and services of over 60 of the
world's leading implementers, provides a tremendous opportunity to measure
the interoperability of our own SIP-based solutions and of the industry as a
whole."
An early supporter of SIP, Mitel Networks has participated in the previous
six SIPit events. At SIPit13 Mitel Networks will test a range of SIP-based
desktop devices and communications platforms including the 5055 SIP Phone
and the 3050 Integrated Communications Platform.
For registration and information about SIPit13 in Ottawa, please visit:
www.mitel.com/sipit.
About SIP
An Internet Engineering Task Force (IETF) standard, SIP is an open,
Internet-genuine protocol for establishing and managing multi-party,
mixed-media sessions over converged networks. SIP enables the creation and
deployment of feature-rich services that go far beyond simple VoIP calls.
About Mitel Networks
Mitel Networks is a market-leader for voice, video, and data convergence
over broadband networks. With a focus on the user experience, the company
delivers advanced communications solutions that are easily customized for
individual business needs. Through intuitive desktop appliances and
applications, businesses are provided with innovative ways to manage
information and resources. Vertical markets benefit from integrated
solutions that protect existing investments, while enabling new ways to be
more cost effective and productive. Mitel Networks is headquartered in
Ottawa, Canada with offices, partners, and resellers worldwide. For more
information please visit
www.mitel.com
-30-
For more information, please contact:
Jon Carvill
Mitel Networks
613.592.2122
jon_carvill(a)mitel.com
[View Less]
Hi,
I have just updated my source version from CVS and find that I get
hundreds of warnings about incompatible pointer types when doing make all.
Have you changed the module export structure (the warning seem to point to
there)?
thanks
Stephan
--
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 …
[View More]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
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for messages of this kind. Opinions, conclusions and other
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business of the University of Stirling shall be understood as neither
given nor endorsed by it.
[View Less]
Yes, the CSeq is incremented.
Thanks,
-g
>
>Hello Greg,
>
>On 11-03 07:40, Greg Fausak wrote:
>> What is weird about the OPTIONS request is that it is
>> including the established Callid.
>
> That is imho fine as long as CSeq in OPTIONS has higher value than
> CSeq in INVITE.
>
> Jan.
>
>--8GpibOaaTibBMecb
>Content-Type: application/pgp-signature
>Content-Disposition: inline
>
>-----BEGIN PGP SIGNATURE-----
>Version: GnuPG v1.2.…
[View More]1 (GNU/Linux)
>
>iD8DBQE+bfTyA0X9pGsVw/URAirmAJ9iqtJFrHLYmUm2rMY7PL0yuvOQgwCdFDXn
>EnqT7XpxTAAArEhhT2p9x7E=
>=QKSs
>-----END PGP SIGNATURE-----
>
>--8GpibOaaTibBMecb--
>
[View Less]
Hi Nils, I've copied my ser.cfg file into the body of this email. As you can
see 'fork=yes'. I've also copied the output of the serctl ps. As you can see
my config is pretty basic at this point. Thansk Mike
0 8512 attendant
1 8514 receiver child=0 sock=0 @ 127.0.0.1::5060
2 8515 receiver child=1 sock=0 @ 127.0.0.1::5060
3 8516 receiver child=2 sock=0 @ 127.0.0.1::5060
4 8517 receiver child=3 sock=0 @ 127.0.0.1::5060
5 8518 receiver …
[View More]child=0 sock=1 @ 10.100.100.243::5060
6 8519 receiver child=1 sock=1 @ 10.100.100.243::5060
7 8520 receiver child=2 sock=1 @ 10.100.100.243::5060
8 8521 receiver child=3 sock=1 @ 10.100.100.243::5060
9 8522 fifo server
10 8539 timer
#
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (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 ----------------------------------
# Uncomment this if you want to use SQL database
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "//usr/lib/ser/modules/auth.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth", "secret", "alsdkhglaksdhfkloiwr")
modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth", "password_column", "password")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==0, or excessively long requests
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");
};
# Do strict routing if pre-loaded route headers present
rewriteFromRoute();
# 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=~"to-talk.com") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("to-talk.com", "subscriber")) {
www_challenge("to-talk.com", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
> Hello Mike,
>
> On Tuesday 11 March 2003 14:28, mjkane(a)attbi.com wrote:
> > Hello, I have sucessfully configured SER to connect to mysql, but, now when
> > I execute ./ser start only 3 process are loaded. If I load SER using the
> > default ser.cfg file it loads 11 processes. Any thoughts..
>
> sounds like you configured 'fork=no'. Then you would have one timer process,
> one process for fifo, and one answering requests. You can check this with
> 'serctl ps'.
>
> Greetings
> Nils
[View Less]
Hello, I have sucessfully configured SER to connect to mysql, but, now when I
execute ./ser start only 3 process are loaded. If I load SER using the
default ser.cfg file it loads 11 processes. Any thoughts..
Mike
If I have a SIP message in a char* variable, then I assume that I can parse
it into a struct sip_msg* using a function in ser. The most obvious function
seems to be parse_msg(). This does not seem to work to my expectation.
Here is a code snippet (which is close to the code in receive.c):
char* response // contains my message
struct sip_msg msg // this is were the result should go
memset(msg, 0, sizeof(struct sip_msg));
response[len] = 0;
msg->buf = response;
msg->len = len;
if (…
[View More]parse_msg(response, len, msg))!=0) {
LOG(L_ERR, "ERROR");
}
LOG(L_ERR, "After parse msg ...")
dump_sip_msg(msg) // this just prints the fields of the message to LOG
dump_sip_msg() works fine on the incoming sip message, so the problem is
either in my SIP message or in my use of the parse function. However, I
think my message is fine (it is the received message with the first line
replaced with SIP/2.0 600 Policy Server Msg date time). I enclose a cut from
/var/log/messages.
Stephan
/var/log/messages:
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: After parse_msg...
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: polserv: dump_sip_msg()
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->id: int = 0
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->first_line:
msg_start (rep) 600 Policy Server Msg 2003-03-07 14:52:24^M Via: SIP/2.0/UDP
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: DEBUG: dump_hdr_field:
type=1, name=Via, body=SIP/2.0/UDP 139.153.254.196:5062, parsed=0x80a584c,
next=(nil)
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: DEBUG: dump_hdr_field:
type=1, name=Via, body=SIP/2.0/UDP 139.153.254.196:5062, parsed=0x80a584c,
next=(nil)
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->parsed_flag 1
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: DEBUG: dump_hdr_field:
type=1, name=Via, body=SIP/2.0/UDP 139.153.254.196:5062, parsed=0x80a584c,
next=(nil)
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: DEBUG: dump_hdr_field:
field is NULL
Mar 7 14:52:41 d254196 last message repeated 20 times
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->eoh: (null)
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->unparsed: CSeq: 7635
REGISTER^M To: "Stephan" <sip:mrs@cs.stir.ac.uk>^M Expires: 900^M From:
"Stephan" <sip:mrs@cs.stir.ac.uk>^M Call-Id: 138108089(a)139.153.254.196^M
User-Agent: KPhone/2.11^M Event: registration^M Allow-Events: presence^M
Contact: "root"
<sip:root@139.153.254.196:5062;transport=UDP>;methods="INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"^M Content-Length: 0^M ^M
^M
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->buf: SIP/2.0 600
Policy Server Msg 2003-03-07 14:52:24^M Via: SIP/2.0/UDP
139.153.254.196:5062^M CSeq: 7635 REGISTER^M To: "Stephan"
<sip:mrs@cs.stir.ac.uk>^M Expires: 900^M From: "Stephan"
<sip:mrs@cs.stir.ac.uk>^M Call-Id: 138108089(a)139.153.254.196^M User-Agent:
KPhone/2.11^M Event: registration^M Allow-Events: presence^M Contact: "root"
<sip:root@139.153.254.196:5062;transport=UDP>;methods="INVITE, MESSAGE,
INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"^M Content-Length: 0^M ^M
^M
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->len: 475
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: msg->parsed_uri_ok: int =
0
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: polserv: all fine,
returning now ...
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: ERROR: parse_sip_msg_uri:
bad uri <600>
Mar 7 14:52:41 d254196 /usr/local/sbin/ser[9566]: WARNING: do_action:error
in expression
--
Dr Stephan Reiff-Marganiec
Research Fellow
Department of Computing Science; University of Stirling
email: srm(a)cs.stir.ac.uk tel: 01786 46 7448
--
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. Opinions, conclusions and other
information in this message that do not relate to the official
business of the University of Stirling shall be understood as neither
given nor endorsed by it.
[View Less]
When I tryto go the page localhost/admin/index.php I get the following
error.
Warning: mysql_pconnect() [function.mysql-pconnect
<http://www.php.net/function.mysql-pconnect>]: Can't connect to local MySQL
server through socket '/tmp/mysql.sock' (2) in
/usr/local/serweb/phplib/db_mysql.inc on line 73
Database error: pconnect(localhost, ser, $Password) failed.
MySQL Error: ()
Session halted.
Any help would ne greatly appreciated.
Thanks
Craig Wolfe
Hello all, what file directs where SER looks for the mysql.sock file. In my
case mysql.sock is located in /tmp and as you can see SER is lookin for it
in /var/lib/mysql
Mar 10 16:46:33 ttalksvr /usr/sbin/ser[6592]: connect_db(): Can't connect to
local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)
Thanks Mike
We are contemplating building an Admin interface for SER that could run on a
windows os. We like the fact that the server is running on Linux but we
require a more intuitive interface for administration.
A web interface is not acceptable, just not enough functionality. We would
like to create an executable that runs on windows which interfaces with SER
server running on Linux.
I am curious if anyone else could make use of a GUI that would incorporate
all of the SER features and functionality. …
[View More]What do you think? Should we
attempt it? Does anyone have any preference on the language we use, e.g.
JAVA, C++ ?
Erik Lagerway
Xten Networks Inc.
7170-515 West Hastings St
Vancouver, BC
V6B5K3
Ph. 1.604.878.0440 ext.5
[View Less]
Some interesting modifications I have made to improve
acc process.
1)
--------------
Some changes made to acc.c to strip down the user
names for output and also to get missed calls logged
with callee rather than caller. It was showing missed
calls on the callers missed_calls.php which is
incorrect.
Here are the extracts, I can post the full prog if
required:
/*
JF added following block to extract to user
*/
str_copy(&to_user, &rq->to->body);
if ((ul_get_user(&…
[View More]to_user) < 0) ||
!to_user.len) {
LOG(L_ERR, "ERROR: acc_request: Error
while extracting to_username\n");
return -1;
}
/*
str_copy(&user, &rq->first_line.u.request.uri);
JF modified this copy statement to create correct acc
username (from not to!)
*/
str_copy(&user, &rq->from->body);
2)
---------------------
I modified the $q in accounting.php to the below to
include calls where BYE sent by callee as well as
caller (before it was not displaying calls when closed
by callee). Also elimination of double counting of
calls due to
multiple INVITEs being sent by UAC. max(time) is
required as distinct rows are produced if the invites
are > 1sec apart. "Group by" puts all the
INVITEs together so that the latest one can be
selected with max(). Assume that latest invite is
correct for measuring call start time. The max(time)
column has been aliased to c2 so that it can be used
in the output $time=Substr($row->c2,0,16).
Here it is:
$q="select distinct t1.sip_callid, max(t1.time) c2,
t1.sip_to,
sec_to_time(unix_timestamp(t2.time)-unix_timestamp(max(t1.time)))
as length ".
"from ".$config->table_accounting." t1,
".$config->table_accounting." t2 ".
"where (t1.user='".$auth->auth["uname"]."' and
t1.sip_method='INVITE') and
((t2.user='".$auth->auth["uname"]."' or
t2.sip_to='".$auth->auth["uname"]."') and
t2.sip_method='BYE') and ".
"t1.sip_callid=t2.sip_callid group by t1.sip_callid
".
"order by t1.time desc";
3) I modified the &q in missed_calls.php to the below
to include calls to aliases as well. The query looks
in the missed_calls table for the sip_to = logged-in
user and also joins with aliases table to extract
calls to alias of contact. Otherwise missed_calls.php
only provided calls made directly to a user and
ignored calls to aliases.
Here it is:
$q="select distinct t1.user, t1.sip_to, t1.time,
t1.sip_status, t1.sip_callid from
".$config->table_missed_calls."
t1,".$config->table_aliases." t2 where
t1.sip_to='".$auth->auth["uname"]."' OR
('sip:".$auth->auth["uname"]."@".$config->default_domain."'=t2.contact
AND t2.user=t1.sip_to) order by time desc";
echo $q;
4)
-----------------
Any comments will be appreciated.
__________________________________________________
Do you Yahoo!?
Yahoo! Tax Center - forms, calculators, tips, more
http://taxes.yahoo.com/
[View Less]
How about a java applet for use with a web browser? I'd have no problem
with a
requirement for a Windows PC for management, but I can see it being an
issue
for others.
And yes I would use it, in whatever form it took.
Dan
-----Original Message-----
From: Erik Lagerway [mailto:sipdev@xten.com]
Sent: Thursday, March 06, 2003 1:50 PM
To: sip-implementors(a)cs.columbia.edu
Cc: serusers(a)lists.iptel.org
Subject: [Serusers] Admin Interface
We are contemplating building an Admin …
[View More]interface for SER that
could run on a windows os. We like the fact that the server is running
on Linux but we require a more intuitive interface for administration.
A web interface is not acceptable, just not enough
functionality. We would like to create an executable that runs on
windows which interfaces with SER server running on Linux.
I am curious if anyone else could make use of a GUI that would
incorporate all of the SER features and functionality. What do you
think? Should we attempt it? Does anyone have any preference on the
language we use, e.g. JAVA, C++ ?
Erik Lagerway
Xten Networks Inc.
7170-515 West Hastings St
Vancouver, BC
V6B5K3
Ph. 1.604.878.0440 ext.5
[View Less]
Hi,
I tried to use the SER SIP server together with a number of different
UAs. I successfully used a Pingtel phone, pingtel softphone, the
Ubiquity UA, and kphone on linux. However, when I tried to use SIPC from
Columbia Uni I keep getting eror messages 400: Bad Request. As all the
other UAs are working fine I assume this may be a problem with sipc.
Have you any experiences with using sipc and ser?
I attach an error log between the Pingtel phone (139.153.254.222) is
trying to send an …
[View More]INVITE to sipc (139.153.254.34). SER (and ngrep) are
running on 139.153.254.50. (registrations are going through fine and
also invites the other way (from sipc to pingtel) go through ok).
Thanks very much for your help!
Regards,
Mario
--
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
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. Opinions, conclusions and other
information in this message that do not relate to the official
business of the University of Stirling shall be understood as neither
given nor endorsed by it.
[View Less]
I everybody.
where can i download serweb????
at http://developer.berlios.de/projects/serweb/ there is not any release for
download and in home page it says "Web-based user provisioning, serweb,
available"
thank's all
-------------------------------------------------
This mail sent through IMP: http://mail.info.unlp.edu.ar/
Jiri,
Scenario is providing IP Telephony to the household.
I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system).
Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a …
[View More]different location when owner is not around and make a 'free' call, return back the phone and the billing still charge the original subsriber.
Any other suggestion to counter this issue is much appreacited.
SSng
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
Sent: Wednesday, March 05, 2003 12:18 AM
To: Ng, Soo Sim; serusers(a)lists.iptel.org
Subject: Re: [Serusers] multiple registration on one user login
At 03:08 PM 3/4/2003, Ng, Soo Sim wrote:
>I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons.
>
>Is there a way to achieve this requirement with SER?
If that is your desparate wish, it is little overhead to make you happy.
I'm still not sure though, it is a useful thing.
Maybe an operator can make more revennues if my wife can accept calls at
any phone in my building and initiate calls in parallel with my doughter.
What are exactly the billing/security reasons here?
-Jiri
[View Less]
Hello,
I fear that such a case can't be avoided with allowing only
a single registration. If I steal your phone away from your
desk, you will not register with it anymore, but I will and
we will have exactly one valid registration. Leaving SIP
phones with hard-wired passwords on your desk has simply the
same potential as leaving your credit-card or cell-phone there.
What can be done about fraud?
User education -- don't leave your money and phone unattended.
Hotline -- report stolen phones …
[View More]to lock the account.
PIN Lock -- use phones which can log-off and log-on (I'm not aware
of any now -- only 3com used to do that)
-Jiri
ps -- ability to move is a feature. I know people who are very glad
to use Vonage's US phone number and move with their ATAs and the
US phone number around in Europe.
At 11:37 PM 3/5/2003, Ng, Soo Sim wrote:
>Jiri,
>
>Scenario is providing IP Telephony to the household.
>I am more concern about the security of the Hardphone. I am thinking of auto-provisioned the hardphone (eg C7960, ATA186) without subsriber intervention. What the subscriber know is their phone # (Just like legacy phone system).
>
>Since the Hardphone is 'hard-coded', the phone can move round the vicinity of the redisential area and still able to make a call. Potentially this will lead to abuse, as someone may take the phone to a different location when owner is not around and make a 'free' call, return back the phone and the billing still charge the original subsriber.
>
>Any other suggestion to counter this issue is much appreacited.
>
>SSng
>
>-----Original Message-----
>From: Jiri Kuthan [mailto:jiri@iptel.org]
>Sent: Wednesday, March 05, 2003 12:18 AM
>To: Ng, Soo Sim; serusers(a)lists.iptel.org
>Subject: Re: [Serusers] multiple registration on one user login
>
>
>At 03:08 PM 3/4/2003, Ng, Soo Sim wrote:
>>I have such requirements. In providing sip-based residential ip telephony, I would like to restrict each home subsriber is only allowed to register one UA per account. This would make easy for billing purposes and for security reasons.
>>
>>Is there a way to achieve this requirement with SER?
>
>If that is your desparate wish, it is little overhead to make you happy.
>I'm still not sure though, it is a useful thing.
>
>Maybe an operator can make more revennues if my wife can accept calls at
>any phone in my building and initiate calls in parallel with my doughter.
>
>What are exactly the billing/security reasons here?
>
>-Jiri
--
Jiri Kuthan http://iptel.org/~jiri/
[View Less]
I have to dig at it abit, but it may also be a codec issue on the Phone.
I had a similar error before telling the Cisco which codec to use. I see
you are using G711Alaw. Can you try G711Ulaw?
The PSTN hand-off section of your config looks very familiar to me, so if
you did pull it from the Howto, note that it is an example of what worked
for me. I've seen quite a few much more sophisticated scripts on the list.
Dan
-----Original Message-----
From: Jiri Kuthan [mailto:jiri@iptel.org]
…
[View More]Sent: Wednesday, March 05, 2003 2:29 AM
To: Rikard Westlund; serusers(a)lists.iptel.org
Subject: Re: [Serusers] Messenger 4.7, CIsco and PSTN
I suspect what happens is that you forward the requests with your server's address in its r-uri to gateway "as is" and the Cisco
gateway would like to see its IP address in the r-uri instead. Try rewriting r-uri -- see bellow.
As for the Messenger problem, see our doc http://www.iptel.org/ser/doc/seruser-html/x878.html#AEN890
-Jiri
At 11:04 AM 3/5/2003, Rikard Westlund wrote:
>Hi all,
>
>I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3.
>
>As clients I use Pingtel and messenger 4.7. I have followed the setup
>guide on http://www.fitawi.com/ser-Howto.html
>
>I can register the pingtel phone with no problem. I can call from the
>PSTN to the pingtel via a Cisco AS5300 with no problems.
>
>When i try toi call from pingtel to PSTN iget the following answer:
>
>1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip 2. from
>ser to pingtel - Status: 100 trying 3. from ser to cisco - INVITE
>sip:<pstnnumber>@serserver_ip 4. from cisco to ser - Status: 400 bad
>request - ínvalid IP address' 5. from cisco to ser - Status: 400 bad
>request - ínvalid IP address'
>
>This is my ser.cfg:
>
># $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
>#
># simple quick-start config script
>#
>
># ----------- global configuration parameters ------------------------
>
>debug=4 # debug level (cmd line: -dddddddddd)
>fork=yes
>log_stderror=no # (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 ----------------------------------
>
># Uncomment this if you want to use SQL database
>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"
>
># Uncomment this if you want digest authentication
># mysql.so must be loaded !
>loadmodule "//usr/lib/ser/modules/auth.so"
>
># ----------------- setting module-specific parameters ---------------
>
># -- usrloc params --
>
>#modparam("usrloc", "db_mode", 0)
>
># Uncomment this if you want to use SQL database
># for persistent storage and comment the previous line
>modparam("usrloc", "db_mode", 2)
>
># -- auth params --
># Uncomment if you are using auth module
>#
>#modparam("auth", "secret", "alsdkhglaksdhfkloiwr") modparam("auth",
>"calculate_ha1", yes) #
># If you set "calculate_ha1" parameter to yes (which true in this config),
># uncomment also the following parameter)
>#
>modparam("auth", "password_column", "password")
>
># ------------------------- request routing logic -------------------
>
># main routing logic
>
>route{
>
> # initial sanity checks -- messages with
> # max_forwars==0, or excessively long requests
> 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;
> };
>
>
> # Do strict routing if pre-loaded route headers present
> rewriteFromRoute();
>
> # 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") {
>
># Uncomment this if you want to use digest authentication
> if (!www_authorize("norrtull.nexus.se", "subscriber")) {
> www_challenge("norrtull.nexus.se", "0");
> break;
> };
>
> save("location");
> break;
> };
>
># attempt handoff to PSTN
>
> if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This assumes that the caller is
> log("Forwarding to PSTN\n"); ## registered in our realm
*** here *** rewrite uri prior to fwd-ing.
rewritehostport("cisco_ip:5060");
> t_relay_to( "cisco_ip", "5060"); ## Our Cisco router
> break;
> };
>
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> break;
> };
> };
> # forward to current uri now
> if (!t_relay()) {
> sl_reply_error();
> };
>
>}
>
>---------------------------------
>
>In the cisco I have the following config:
>
>!
>dail-peer voice 25 voip
>destination-pattern XXXX
>session protocol sipv2
>codec g711alaw
>no vad
>session target ipv4:serserver_ip
>!
>
>I have added 2 subscribers with the serctl command and registration is
>working well from pingtel. In Messenger 4.7 it's not working at all. I
>get 401 Unauthorized.
>
>Well I think thats about it..
>
>Please feel free to contact me if you need more information
>
>Best regards
>
>Rikard Westlund
>
>
>
>_________________________________________________________________
>Tired of spam? Get advanced junk mail protection with MSN 8.
>http://join.msn.com/?page=features/junkmail
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
[View Less]
Hi all,
I have a Ser 0.8.10-2 install on a Redhat 7.3 kernel 2.4.18-3.
As clients I use Pingtel and messenger 4.7. I have followed the setup guide
on
http://www.fitawi.com/ser-Howto.html
I can register the pingtel phone with no problem. I can call from the PSTN
to the pingtel via a Cisco AS5300 with no problems.
When i try toi call from pingtel to PSTN iget the following answer:
1. from pingtel to ser - INVITE sip:<pstnnumber>@serserver_ip
2. from ser to pingtel - Status: 100 …
[View More]trying
3. from ser to cisco - INVITE sip:<pstnnumber>@serserver_ip
4. from cisco to ser - Status: 400 bad request - ínvalid IP address'
5. from cisco to ser - Status: 400 bad request - ínvalid IP address'
This is my ser.cfg:
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (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 ----------------------------------
# Uncomment this if you want to use SQL database
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "//usr/lib/ser/modules/auth.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth", "secret", "alsdkhglaksdhfkloiwr")
modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth", "password_column", "password")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==0, or excessively long requests
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;
};
# Do strict routing if pre-loaded route headers present
rewriteFromRoute();
# 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") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("norrtull.nexus.se", "subscriber")) {
www_challenge("norrtull.nexus.se", "0");
break;
};
save("location");
break;
};
# attempt handoff to PSTN
if (uri=~"^sip:1[0-9]*@norrtull.nexus.se") { ## This
assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in
our realm
t_relay_to( "cisco_ip", "5060"); ## Our Cisco
router
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
# forward to current uri now
if (!t_relay()) {
sl_reply_error();
};
}
---------------------------------
In the cisco I have the following config:
!
dail-peer voice 25 voip
destination-pattern XXXX
session protocol sipv2
codec g711alaw
no vad
session target ipv4:serserver_ip
!
I have added 2 subscribers with the serctl command and registration is
working well from pingtel. In Messenger 4.7 it's not working at all. I get
401 Unauthorized.
Well I think thats about it..
Please feel free to contact me if you need more information
Best regards
Rikard Westlund
_________________________________________________________________
Tired of spam? Get advanced junk mail protection with MSN 8.
http://join.msn.com/?page=features/junkmail
[View Less]
I copied a few SNOM phone messages to the list.
I am having problems communicating with the
SNOM phone.
I guess it isn't working because the SNOM phone
sends a PRACK after the 183, and the received
OK doesn't have a 'Record-Route'. This doesn't make
sense to me. I don't know why the phone is listening
to the first OK (the OK to the PRACK) rather than the
second OK (the OK to the INVITE).
If that is the way it is then do I have the PROXY mis-configured?
-----
if(method=="INVITE" | …
[View More]method=="BYE" | method=="PRACK")
{
log(1,"TRACE: addRecordRoute()");
setflag(1);
setflag(2);
addRecordRoute();
};
append_hf("P-hint: ATEND\r\n");
if(!t_relay())
{
sl_reply_error();
break;
};
When I do a ngrep trace, I see the packet being relayed...
----
#
U 2003/03/03 20:05:59.858473 216.87.145.22:5060 -> 216.87.144.203:5060
PRACK sip:2143357976@216.87.144.203;branch=0 SIP/2.0.
Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg.
Max-Forwards: 70.
RAck: 5620 2 INVITE.
From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i.
To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C.
Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22.
CSeq: 4 PRACK.
Route: <sip:92143357976@216.87.144.196:5060;user=phone>.
Contact: <sip:4695461245@216.87.145.22:5060;line=1>.
Content-Length: 0.
Authorization: Digest
username="4695461245",realm="augustvoice.net",nonce="3e640b3300000000dbb
edccd04f755cac7e3b6dd55d202c9",uri="sip:",response="3693b828395cd40fe1b1
ab5f9ad61308",algorithm=md5.
.
#
U 2003/03/03 20:05:59.859540 216.87.144.203:5060 -> 216.87.144.196:5060
PRACK sip:92143357976@216.87.144.196:5060;user=phone SIP/2.0.
Record-Route: <sip:2143357976@216.87.144.203;branch=0>.
Via: SIP/2.0/UDP 216.87.144.203;branch=z9hG4bKe44a.ff0091c4.0.
Via: SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg.
Max-Forwards: 69.
RAck: 5620 2 INVITE.
From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i.
To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C.
Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22.
CSeq: 4 PRACK.
Contact: <sip:4695461245@216.87.145.22:5060;line=1>.
Content-Length: 0.
Authorization: Digest
username="4695461245",realm="augustvoice.net",nonce="3e640b3300000000dbb
edccd04f755cac7e3b6dd55d202c9",uri="sip:",response="3693b828395cd40fe1b1
ab5f9ad61308",algorithm=md5.
P-hint: ATEND.
.
#
U 2003/03/03 20:05:59.861477 216.87.144.196:5060 -> 216.87.144.203:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
216.87.144.203;branch=z9hG4bKe44a.ff0091c4.0,SIP/2.0/UDP
216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg.
From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i.
To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C.
Date: Tue, 04 Mar 2003 02:05:59 GMT.
Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 4 PRACK.
Content-Length: 0.
.
#
U 2003/03/03 20:05:59.861705 216.87.144.203:5060 -> 216.87.145.22:5060
SIP/2.0 200 OK.
Via:SIP/2.0/UDP 216.87.145.22:5060;branch=z9hG4bK-bec8z5qwsbqg.
From: "snom man" <sip:4695461245@augustvoice.net>;tag=1r1bpkl72i.
To: <sip:2143357976@augustvoice.net;user=phone>;tag=63631E34-197C.
Date: Tue, 04 Mar 2003 02:05:59 GMT.
Call-ID: 3c26f8de39d9-7dyvkrekeha3(a)216.87.145.22.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 4 PRACK.
Content-Length: 0.
.
-----
There is a Record-route going out, but not one coming back.
Does a Record-route need to be injected into the last
OK back to the UA?
This is too hard :-)
---greg
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