Hi Group,
Can someone tell me the real difference between SER and ASTERISK ?
Are they like C++ and C, a CAR and a PLANE, ...
Are they different by their purpose, or by their nature? Do they serve the
same purpose? ... ...?
I am a confused guy on that matter.
Thanks,
__________________________________
NZEYIMANA Emery Fabrice
NEFA Computing Services, Inc.
P.O. Box 5078 Kigali
Office Phone: +250-51 11 06
Office Fax: +250-50 15 19
Mobile: +250-08517768
Email: dg(a)nefacomp.net
http://www.nefacomp.net/
Hi,
I have installed
ser-0.8.11,
FreeRADIUS-0.9.0,
radiusclient-0.3.2.
I can register two useragents(Kphone) to ser with default authorization ( www_authorize())via mysql database.
But when I use radius_www_authorize(),the useragent(Kphone)dostnot get authorized.
What I have to do further to get radius authentication.
Please help me
Thanks in advance
Regards
Xten_india.
I get the following message in radius.log file.
------------------------------------------------
Sat Sep 13 13:29:48 2003 : Info: Listening on IP address 192.168.100.101, ports 1812/udp and 1813/udp, with proxy on 1814/udp.
Sat Sep 13 13:29:48 2003 : Info: Ready to process requests.
Sat Sep 13 13:30:04 2003 : Error: rlm_eap: EAP-Message not found
Sat Sep 13 13:30:04 2003 : Auth: Login OK: [root/support] (from client ser port 1645)
--------------------------------------------------
My configuration file is
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/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)
# -- auth params --
# Uncomment if you are using auth module
#
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
modparam("auth_radius", "radius_config",
"/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_db", "db_url","sql://root:support@localhost/radius")
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")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("acc", "log_level", 1)
#modparam("acc", "radius_flag", 1)
modparam("acc", "log_flag", 1)
modparam("auth_radius", "service_type", 15)
# ------------------------- 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;
};
# 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") {
# Uncomment this if you want to use digest authentication
if (!radius_www_authorize("localdomain.com")) {
www_challenge("localdomain.com", "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; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
---------------------------------
Do you Yahoo!?
Yahoo! SiteBuilder - Free, easy-to-use web site design software
Thank to courtesy of Jeff Pulver, there will be a SIP/SER workshop
hosted in the upcoming VoN in Boston, on Monday 22nd 5-6.30 pm, after
the FWD meeting.
I'm very glad to invite you to this event. All you need to do to
participate is to register for the FWD meeting and get a temporary
badge for conference center access. To register, please visit:
<http://pulver.com/fwdugm/register.html>
In this workshop, technologists with experience in integrating
SIP services powered by SER technology will gather and share
their hands-on experience in compact, in-depth presentations
and follow-up open-mike discussion:
- Jiri Kuthan of iptel.org will give a status update on SER
- Ed Guy of pulver.com will report on infrastructure powering FWD
- Jeremy George of Yale University will present on integration
with SIP-based presence services
- Michael Haberler will report on the at43 public SIP platform in Austria
- Greg Fausak of Addaline.com will report on their SIP VoIP
operation
- Johnson Wu of Jasomi will present on NAT traversal scenarios
- David Li of Grandstream will show up with a story on their SIP phones.
-Jiri
Thank to courtesy of Jeff Pulver, there will be a SIP/SER workshop
hosted in the upcoming VoN in Boston, on Monday 22nd 5-6.30 pm, after
the FWD meeting.
I'm very glad to invite you to this event. All you need to do to
participate is to register for the FWD meeting and get a temporary
badge for conference center access. To register, please visit:
<http://pulver.com/fwdugm/register.html>
In this workshop, technologists with experience in integrating
SIP services powered by SER technology will gather and share
their hands-on experience in compact, in-depth presentations
and follow-up open-mike discussion:
- Jiri Kuthan of iptel.org will give a status update on SER
- Ed Guy of pulver.com will report on infrastructure powering FWD
- Jeremy George of Yale University will present on integration
with SIP-based presence services
- Michael Haberler will report on the at43 public SIP platform in Austria
- Greg Fausak of Addaline.com will report on their SIP VoIP
operation
- Johnson Wu of Jasomi will present on NAT traversal scenarios
- David Li of Grandstream will show up with a story on their SIP phones.
-Jiri
/* sorry for double-posting, I realized in previous email I dropped subject line */e
--
Jiri Kuthan http://iptel.org/~jiri/
Thank to courtesy of Jeff Pulver, there will be a SIP/SER workshop
hosted in the upcoming VoN in Boston, on Monday 22nd 5-6.30 pm, after
the FWD meeting.
I'm very glad to invite you to this event. All you need to do to
participate is to register for the FWD meeting and get a temporary
badge for conference center access. To register, please visit:
<http://pulver.com/fwdugm/register.html>
In this workshop, technologists with experience in integrating
SIP services powered by SER technology will gather and share
their hands-on experience in compact, in-depth presentations
and follow-up open-mike discussion:
- Jiri Kuthan of iptel.org will give a status update on SER
- Ed Guy of pulver.com will report on infrastructure powering FWD
- Jeremy George of Yale University will present on integration
with SIP-based presence services
- Michael Haberler will report on the at43 public SIP platform in Austria
- Greg Fausak of Addaline.com will report on their SIP VoIP
operation
- Johnson Wu of Jasomi will present on NAT traversal scenarios
- David Li of Grandstream will show up with a story on their SIP phones.
-Jiri
--
Jiri Kuthan http://iptel.org/~jiri/
hi guys,
has anyone of you done any interop with audiocodes fxs?
i've tried 2 firmware versions but i still get sip stack errors.
like this
Sep 16 11:42:47 10.17.3.1 ( lgr_stk_mngr)( 330)!! [ERROR] StackMngr: Message was not Parsed correctly ! Last parsed line was :MaxForwardsHeader:
Sep 16 11:42:47 10.17.3.1 ( lgr_stk_mngr)( 331)!! [ERROR] The error was = Unexpected '>'
Sep 16 11:42:47 10.17.3.1 ( lgr_stk_mngr)( 332)!! [ERROR] The Line of the error was = 3
Sep 16 11:42:47 10.17.3.1 ( lgr_stk_mngr)( 333)!! [ERROR] The Column of the error was = 82
i'm currently stuck, maybe i could get some ideas from you guys
thanks
~kelvin
Let's say I want to allow user1 to register 1234567890(a)iptel.org with a
password of pw1.
Based on the User Guide, it is being worked on in module domain. Based on
the README for module uri, it is implemented there.
Is it actually implemented anywhere? If yes, what has to go in which column
of which table? I tried:
insert into uri values('user1', 'iptel.org' , '1234567890' ,'');
and
insert into uri values('1234567890', 'iptel.org' , 'user1' ,'');
and restarted. Looking at the debug, it looks like it just compared Digest
and To, without any additional checking when that failed.
Thanks.
Hi,
I am new to SER. I want to use SER in my school project. Since SER is written in C, I was wondering if we can add any Java Extensions to it and make it work. Do you provide any API's?
This might be a very silly question. Pardon my ignorance.
Thanks,
Madhuri.
---------------------------------
Do you Yahoo!?
Yahoo! SiteBuilder - Free, easy-to-use web site design software
Hello,
Anybody between you knows how we can realize a call entering of RNIS through
Gateway SIP
Thank you in advance
Cordialement
hassan
-------------------------------------------------
This mail sent through IMP: http://horde.org/imp/
Hello all,
I installed serweb, it works well, but I have a problem at the level of the
application: when I want to make a subscription with an address numeric SIP for
example: 9991(a)domaine.com, it not accepts it , it sends me a message (user
name does not follow suggested convantions),
Another question, if I want to try to make communications between a softphone
SIP and an analogical telephone through a cisco router, it is enough to go the
commands this below into the cisco router, and the others into the server SER:
In the router:
Dial-peer voice 999 voip
Destination-pattern 555999. ** Associate the number arranges 555-9990 to 9999
with our SIP server
Session protocol sipv2 ** Set this dial-peer to uses(wears out) SIP instead of
Cisco protocols
Session target sip-server ** Send the call to our SIP server. See SIP-UA below
Codec g711ulaw ** Set the default codec to 711-Ulaw (common codec between
customers)
!
Sip-ua
Sip-server ipv4:192.168.0.1
In the fileser.cfg :
if (uri=~"^sip:9[0-9]*@mydomain.com") { ## This assumes that the caller is
log("Forwarding to PSTN\n"); ## registered in our realm
t_relay_to( "192.168.0.2", "5060"); ## Our Cisco router
break;
};
thanks you for advance
hassan
-------------------------------------------------
This mail sent through IMP: http://horde.org/imp/