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 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
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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 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
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 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/
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]
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 - Ãnvalid IP address' 5. from cisco to ser - Status: 400 bad
>request - Ãnvalid 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
>
>
>
>_________________________________________________________________
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>
>_______________________________________________
>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
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 - Ãnvalid IP address'
5. from cisco to ser - Status: 400 bad request - Ãnvalid 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
_________________________________________________________________
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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" | 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
I installed ser stable then unstable debs on my 2.2.20 kernel install of
unstable debian.
Errors:
when init.d/ser start:
Starting ser: serToo much shared memory demanded: 33554432
Terminated
SYSLOG:
Mar 5 10:09:19 webserver ser: ERROR: shm_mem_init: could not attach
shared
memo
ry segment: Invalid argument
Mar 5 10:09:19 webserver ser: could not initialize shared memory pool,
exiting.
a friend sent me:
scmctl
from shmctl(2):
Various fields in a struct shmid_ds were shorts under Linux 2.2 and
have become longs under Linux 2.4. To take advantage of this, a
recompilation under glibc-2.1.91 or later should suffice. (The kernel dis-tinguishes
old and new calls by a IPC_64 flag in cmd.)
Do I need kernel 2.4? Any hints? Have I missed some documentation?
Jaime Hemmett
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
