A very elementary question. How do I get a version of serweb that is
compatible with CVS head version of SER? I can't see it on the ftp site and
I'm a cvs bunny so it'll be something like this...
cd /usr/srcexport
CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/serwebcvs logincvs co -r
ummmdunno serwebAny help would be appreciated.RegardsCameron
I do see it in the portage tree but what about the ser-mysql integration
and all the other add-ons. I don't see them in the portage tree. Do I
need them or does ser come with them already included in the package
that comes from the portage tree distribution.
Wercs Communications
Clay Bryan
Network Administrator
Wercs Communications
400 East First
Casper, WY 82601
<http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe
r%2C+WY++82601&country=us>
CBryan(a)wercs.com <mailto:CBryan@wercs.com>
tel:
fax:
mobile:
307-233-8359
307-233-8701
307-258-7371
Add me to your address book...
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Hi all,
I am trying to setup some cdr generation from a ser server
(version 0.8.14) and I am using a perl script like this:
foreach $var (sort(keys(%ENV))) {
$val = $ENV{$var};
if ( ${var} =~ /^SIP_/){
print FP "${var}=\"${val}\"\n";
}
}
executed from ser.cfg, but I need the sip_response_code (2xx if ok).
I can see that it is in the output from xlog, is it possible
to fetch the value from any environment parameter at all?
And how to do that.
/kk
Hello,
the XLOG-Module prints an error if the User-Agend %ua Parameter is
called on a request that does not contain a User-Agent header:
XLOG: xl_get_useragent: ERROR cannot parse User-Agent header
This is probably a bug, because this error is not showing up on a
non-existant Contact-Header, nor on a non-existant To-Tag. Version is
rel_0_9_0 cvs.
With best regards,
Martin Koenig
So Daniel like i understand the problem is my radius configuration,
another thing is that my ATA sending the same stuff, i mean if i will
change the sip server to different one where i installed freeradius
with ser it's working fine.
Daniel where i can start to fix that problem.?
Thank you very much for your time.
On 4/14/05, Alex <alexandergav(a)gmail.com> wrote:
> So Daniel like i understand the problem is my radius configuration,
> another thing is that my ATA sending the same stuff, i mean if i will
> change the sip server to different one where i installed freeradius
> with ser it's working fine.
>
> Daniel where i can start to fix that problem.?
>
> Thank you very much for your time.
>
>
> On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote:
> > The second REGISTER (the block 3) must contains the response to the
> > authentication challenge carried by 401 reply (block 2). That means that
> > the block 3 must contain an Authorization header with authentication
> > credentials computed according to HTTP-Digest authentication mechanism
> > (RFC2617). Also, see the section 22.Usage of HTTP Authentication in SIP
> > RFC3261 for more about user authentication in SIP.
> >
> > Daniel
> >
> > On 04/14/05 13:16, Alex wrote:
> >
> > >Sorry Daniel , i didn't get that, I send here 4 blocks, 1 one is the
> > >register request the 2 is the reply from the server, 3 is the register
> > >request, 4 is the reply from the server. If you can please point me to
> > >the problem. Because like i see the 2 register requests (1,3 blocks)
> > >are the same.
> > >
> > >
> > >On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote:
> > >
> > >
> > >>As you can see, the second REGISTER does not contain the authentication
> > >>credentials (No Authorization header) in response to 401 reply. So,
> > >>either you didn't configure the phone to authenticate or the Grandstream
> > >>HT286 1.0.5.18 is faulty.
> > >>
> > >>Daniel
> > >>
> > >>
> > >>On 04/14/05 12:35, Alex wrote:
> > >>
> > >>
> > >>
> > >>>Daniel thanks
> > >>>btw it's clean installation of Red Hat Enterprise Linux AS release 3
> > >>>ser-08.14 , freeradius-1.2 , radiusclient-4.8
> > >>>
> > >>>i am sending ngrep port 5060
> > >>>i have here 2 requests of register and the replies to register.
> > >>>
> > >>>
> > >>>xxx.xxx.xxx.xxx - sipserverip
> > >>>telephoneip - ip where the call coming from
> > >>>Phonenumber - phone number
> > >>>
> > >>>--------------------------------------------------------------------------------------------------
> > >>>
> > >>>U telephoneip:10739 -> xxx.xxx.xxx.xxx:5060
> > >>> REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
> > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <
> > >>> sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To:
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip
> > >>> :Phonenumber@telephoneip:10000;user=phone>..Call-ID:
> > >>>1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires:
> > >>>3600..User-Agent
> > >>> : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow:
> > >>>INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt
> > >>> h: 0....
> > >>>#
> > >>>U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000
> > >>> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
> > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex"
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To:
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9
> > >>> 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106
> > >>>REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc
> > >>> e="425e3ac34dc9509392435c11fb260f41420049c7"..Server: Sip EXpress
> > >>>router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
> > >>> xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1912
> > >>>req_src_ip=telephoneip req_src_port=10739 in_uri=sip:xxx.xxx.xxx.xxx
> > >>> out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1"....
> > >>>#
> > >>>
> > >>>U telephoneip:10740 -> xxx.xxx.xxx.xxx:5060
> > >>> REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP
> > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <
> > >>> sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To:
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip
> > >>> :Phonenumber@telephoneip:10000;user=phone>..Call-ID:
> > >>>1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires:
> > >>>3600..User-Agent
> > >>> : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow:
> > >>>INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt
> > >>> h: 0....
> > >>>#
> > >>>U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000
> > >>> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
> > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex"
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To:
> > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9
> > >>> 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106
> > >>>REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc
> > >>> e="425e3acb812b5b2e8aa023e3fcffc618dc4cf661"..Server: Sip EXpress
> > >>>router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392
> > >>> xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1885
> > >>>req_src_ip=telephoneip req_src_port=10740 in_uri=sip:xxx.xxx.xxx.xxx
> > >>> out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1"....
> > >>>#
> > >>>
> > >>>
> > >>>tell me if you need something else.
> > >>>
> > >>>
> > >>>On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote:
> > >>>
> > >>>
> > >>>
> > >>>
> > >>>>Could you post the network dump with REGISTER/401/REGISTER messages? I
> > >>>>will take a look to see if something is wrong.
> > >>>>
> > >>>>
> > >>>>On 04/14/05 12:16, Alex wrote:
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>>Digest realm is the same for the register requests.
> > >>>>>furthermore the realm in To tag is correct.
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>Did you mean To URI instead of To tag?
> > >>>>
> > >>>>Daniel
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>
> > >>>>>what else it can be.
> > >>>>>Thanks for any help.
> > >>>>>
> > >>>>>On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote:
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>>Watch the network traffic (ngrep or ethereal on port 5060) and check the
> > >>>>>>realm from 401 is the same as the one from next REGISTER. Also, when
> > >>>>>>you use the empty realm parameter to radius_ww_authorize() and
> > >>>>>>www_challenge(), the realm is taken from To URI.
> > >>>>>>
> > >>>>>>Daniel
> > >>>>>>
> > >>>>>>
> > >>>>>>On 04/14/05 08:08, Alex wrote:
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>>>Hi guys maybe someone can find the problem, i still can't see anything
> > >>>>>>>going to radius authentication. (the radius logs are empty)
> > >>>>>>>
> > >>>>>>>the register request is coming but it's not going to authenticate
> > >>>>>>>through the radius.
> > >>>>>>>Any help will be appreciated.
> > >>>>>>>
> > >>>>>>>here is the debug from ser :
> > >>>>>>>---------------------------------------------------------------------------------------------
> > >>>>>>>14(1036) parse_headers: flags=-1
> > >>>>>>>14(1036) check_via_address(62.219.158.191, 62.219.158.191, 1)
> > >>>>>>>14(1036) DEBUG:destroy_avp_list: destroing list (nil)
> > >>>>>>>14(1036) receive_msg: cleaning up
> > >>>>>>>9(1012) SIP Request:
> > >>>>>>>9(1012) method: <REGISTER>
> > >>>>>>>9(1012) uri: <sip:xxx.xxx.xxx.xxx>
> > >>>>>>>9(1012) version: <SIP/2.0>
> > >>>>>>>9(1012) parse_headers: flags=1
> > >>>>>>>9(1012) Found param type 232, <branch> = <z9hG4bKfc5751413c832e6d>; state=16
> > >>>>>>>9(1012) end of header reached, state=5
> > >>>>>>>9(1012) parse_headers: Via found, flags=1
> > >>>>>>>9(1012) parse_headers: this is the first via
> > >>>>>>>9(1012) After parse_msg...
> > >>>>>>>9(1012) preparing to run routing scripts...
> > >>>>>>>9(1012) REGISTER: Authenticating user
> > >>>>>>>9(1012) parse_headers: flags=4
> > >>>>>>>9(1012) end of header reached, state=9
> > >>>>>>>9(1012) DEBUG: get_hdr_field: <To> [45];
> > >>>>>>>uri=[sip:phonenumber@xxx.xxx.xxx.xxx;user=phone]
> > >>>>>>>9(1012) DEBUG: to body [<sip:phonenumber@xxx.xxx.xxx.xxx;user=phone>
> > >>>>>>>]
> > >>>>>>>
> > >>>>>>>9(1012) parse_headers: flags=4096
> > >>>>>>>9(1012) get_hdr_field: cseq <CSeq>: <103> <REGISTER>
> > >>>>>>>9(1012) DEBUG: get_hdr_body : content_length=0
> > >>>>>>>9(1012) found end of header
> > >>>>>>>9(1012) pre_auth(): Credentials with given realm not found
> > >>>>>>>9(1012) REGISTER: challenging user
> > >>>>>>>9(1012) build_auth_hf(): 'WWW-Authenticate: Digest
> > >>>>>>>realm="xxx.xxx.xxx.xxx",
> > >>>>>>>nonce="425e063022afc1142ed6730d46da41692ff3ed57"
> > >>>>>>>
> > >>>>>>>_______________________________________________
> > >>>>>>>Serusers mailing list
> > >>>>>>>serusers(a)lists.iptel.org
> > >>>>>>>http://lists.iptel.org/mailman/listinfo/serusers
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>>>
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>
> > >>>
> > >>>
> > >
> > >
> > >
> >
>
Hello there,
we use ser to forward incoming call requests to a remote PSTN.
this system works fine if the client has a non-natted public IP
address.However, I can't tell the same thing for the natted clients and
it doesn't work so far.
using nathelper/rtpproxy, the client (x-lite) can make the connection
and ring the phone but picking up the phone on the other end.. hmm,
well, no audio can be heard.
As a matter of fact, x-lite client times out with error 408 "after" I
hear the remote phone rings. it seems like the client side simply
signaling for connection but no data(audio) transfer can be made.
below is my ser.cfg, does anybody have an idea?
Thanks in advance.
Serhan.
# ----------- global configuration parameters ------------------------
#debug=7 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=yes # (cmd line: -E)
# Uncomment these lines to enter debugging mode
#debug=7
#fork=yes
#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 ----------------------------------
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.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/xlog.so"
#Database URL is here!
modparam("usrloc", "db_url", "sql://ser:pass123@localhost/ser")
#nathelper params.
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 seconds
modparam("nathelper", "ping_nated_only", 1) # Ping only clients
behind NAT
#mysql
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
#modparam("usrloc", "db_mode", 0)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
# main routing logic
alias="domain.com"
alias="sipgw.domain.com"
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;
};
# for testing purposes, simply okay all REGISTERs
if (method=="REGISTER") {
log(1, "REGISTER received\n");
} else {
log(1, "non-REGISTER received\n");
};
if (uri=~"sip:.*[@:]sipgw.domain.com") {
log(1, "request for sipgw.domain.com received\n");
} else {
log(1, "request for other domain received\n");
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entitie
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# PSTN Forwarding
if (uri=~"^sip:1")
{
strip(1);
# forward(pstngw.jondoe.com,5060);
force_rtp_proxy();
t_relay_to_udp("pstngw.jondoe.com", "5060");
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "not exists!");
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();
};
}
Hello list:
Did you guys try Teliann 3100 series phone?
I am having some issue with domain name, auto answer,
proxy_authorization and stun.
If you anybody tried this phone, please share your experience. I will
really appreciate.
Thanks,
Mohammad Khan
Hi,
I am migrating from 0.8.14 to 0.9.0
can anyone tell me how to compile in mysql for ser 0.9.0
first I found that mysql.so was not compiled in standard then
I modified the Makefile and included mysql , nevertheless
ser displays an error.
Apr 18 20:04:20 /usr/local/sbin/ser[4734]: Maxfwd module- initializing
Apr 18 20:04:20 /usr/local/sbin/ser[4734]: bind_dbmod: Module sql does
not export db_use_table function
Apr 18 20:04:20 /usr/local/sbin/ser[4734]: ERROR: mod_init(): Can't bind
database module
Apr 18 20:04:20 /usr/local/sbin/ser[4734]: init_mod(): Error while
initializing module usrloc
thanks
Paul van Schagen
Hi, ALL:
How to limit the maxinum register of UACs and the maxinum current calls of UACs?
I have no idea to limit these from ser.cfg. Does anybody know how to
limit them?
--
Best Regards
Charles
