The PUBLISH request works when using the following configurations(no XCAP):
loadmodule "sl.so"
loadmodule "mysql.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser")
modparam("presence_xml", "force_active", 1)
modparam("presence_xml", "pidf_manipulation", 1)
It doesn't if starting XCAP by updating these, the server can run properly, but cann't handle PUBLISH.
loadmodule "xcap_client.so"
modparam("presence_xml", "xcap_table", "xcap_xml")
modparam("presence_xml", "force_active", 0)
modparam("presence_xml", "xcap_server", "localhost:5065")
I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server?
Thank You!
Kevin
_________________________________________________________________
Windows Live Spaces 中最年轻的成员！
http://miaomiaogarden2007.spaces.live.com/
Hi,
Im trying to compile mi_xmlrpc module in openser but could not succeed. The
following errors are displayed
mi_xmlrpc.c: In function 'xmlrpc_process':
mi_xmlrpc.c:161: error: 'xmlrpc_server_abyss_rpc2_handler' undeclared (first
use in this function)
mi_xmlrpc.c:161: error: (Each undeclared identifier is reported only once
mi_xmlrpc.c:161: error: for each function it appears in.)
mi_xmlrpc.c:167: error: 'xmlrpc_server_abyss_default_handler' undeclared
(first use in this function)
Im using Suse 10.1; The libraries xmlrpc-c-0.9.10-36, libxml-1.8.17-385,
libxml2-devel-2.6.23-13, xmlrpc-c-devel-0.9.10-36 are properly installed.
Few discussions on the mailing list concluded that the module mi_xmlrpc
compiles well on Debian without any dependency issues. Is it true??
Can somebody tell me whether Im missing any dependencies?? Please let me
know how to successfully compile this module.
Thanks and regards,
Purna Chandar M
Hi to all,
i have openser 1.2.1 active from some months.. Suddenly on yesterday the
natping stop to work.
I have 140 contacts in user location almost all of them natted.
There is some limit to user in location table at which the ping stops or
a particular server load?
There is a way to have something in the logs about it??
I didn't found anything in the logs.
This is my config:
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
With a restart of openser all come back to work but i'm afraid if it
will happen again?
Thanks for help,
Bye,
Marcello
Hi,
Arrggghh .. that's one of my attempts to eliminate the broken "BYE"
problem... that's ngrep was captured when I set "modparam("rr",
"enable_double_rr", "0");",
I've paste another ngrep to http://pastebin.ca/674450, this time the
double RR header is enabled.
And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt
the post is "openser").
Even though double RR header is enabled, but for BYE it's still doesn't
process properly :(
For the .cfg file line #130 onward, I did tried t_relay, forward and
force_send_socket,
but none of this will do the trick (force_send_socket was complaining
TLS error due to missing certificate (?) )
Would appreciate if anyone could enlighten me why is this happen ?
Thanks,
David Loh
Klaus Darilion wrote:
> But the INVITE you posted at http://pastebin.ca/673392 also has only
> one Record-Route header.
>
> regards
> klaus
>
> David Loh schrieb:
>> Hi,
>>
>> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK
>> ...but when asterisk disconnected the call and send BYE back to OpenSER,
>> the TLS RR header wasn't present, the only 2 RR header was
>> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" ....
>> I'm puzzled ... is there any command to 'fix' this?
>>
>>
>> Regards,
>> David Loh
>>
>> Klaus Darilion wrote:
>>> The openser proxy should add 2 record-route header (TLS and UDP =
>>> double record route). This is why it does not work.
>>>
>>> regards
>>> klaus
>>>
>>> David Loh schrieb:
>>>> Hi All,
>>>>
>>>> Greeting.
>>>>
>>>> I've been struggle with OpenSER TLS implementation for more than a
>>>> week, since I've ported from UDP to TLS, everything work fine
>>>> except the "BYE" request from Asterisk (loose route), my
>>>> implementation was something like below:
>>>>
>>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk]
>>>>
>>>> My OpenSER.cfg already configured to listen on two port which is :-
>>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or
>>>> even voicemail) having no problem,
>>>> but when the callee disconnect the call, caller will never get hang
>>>> up :(
>>>>
>>>> I've attached my ethereal trace/ngrep to pastebin,
>>>> http://pastebin.ca/673392
>>>>
>>>> Wondering if anyone can help me with the broken "BYE" that returned
>>>> from Asterisk ?
>>>> Line #131, supposedly this line should have contain 2 Via header,
>>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS",
>>>> but somehow the TLS via header was gone !! (compare to previous ACK
>>>> (Line #117) /INVITE (Line #51).
>>>> Due to the missing TLS via header, OpenSER log file was complaining
>>>> "protocol/port mis-match".
>>>>
>>>> The last BYE request (Line #256) is actually firing from Client,
>>>> which contain the "TLS" via.
>>>>
>>>>
>>>> I've even tried "force_send_socket" to port 5061 (instead of 5060)
>>>> from loose route, but it complaining TLS certificate error,
>>>> since Asterisk doesn't support TLS natively, I've no clue why is
>>>> the ACK/INVITE/CANCEL work but not BYE.
>>>> if (loose_route) {
>>>> ....
>>>> if(is_method("BYE")) { force_send_socket(IP:5061); }
>>>> }
>>>>
>>>>
>>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk
>>>> setup,
>>>> I'm really appreciate if you can share your experience with me, or
>>>> pin point what's the mistakes I made here.
>>>>
>>>> Thanks in advance.
>>>>
>>>> Regards,
>>>> David Loh
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users(a)openser.org
>>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>
Hi,
I have an Ubuntu 6.06 install with PostgreSQL 8.1 and OpenSER 1.2.2
which I am testing at the moment.
I have tested OpenSER with MySQL and it worked fine but when when I want
to switch to PostgreSQL I'm having some difficulty.
I have used the postgressqldb.sh script to initialise the database and
it has completed without error.
When I start OpenSER I get the following lines in my PostgreSQL logs:
LOG: connection received: host=127.0.0.1 port=37610
LOG: connection authorized: user=openser database=openser
LOG: SSL SYSCALL error: EOF detected
LOG: could not receive data from client: Connection reset by peer
LOG: unexpected EOF on client connection
I have looked at OpenSER's logs with debugging turned on but I don't see
anything that looks like an error but I would be happy to provide them.
`openserctl moni` returns:
ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo
ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name",
"/tmp/openser_fifo")' in your config
ERROR: and also have loaded the mi_fifo module.
My openser.cfg looks like this:
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
children=4
listen=192.168.20.151
dns=no
rev_dns=no
port=5060
mpath="/usr/local/lib/openser/modules/"
loadmodule "mysql.so"
loadmodule "postgres.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "db_url",
"postgres://openser:openserrw@localhost/openser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "db_url",
"postgres://openser:openserrw@localhost/openser")
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
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");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# 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
if (!method=="REGISTER")
record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
# if you have some interdomain connections via TLS
#if(uri=~"@tls_domain1.net") {
# t_relay("tls:domain1.net");
# exit;
#} else if(uri=~"@tls_domain2.net") {
# t_relay("tls:domain2.net");
# exit;
#}
route(1);
};
# 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("host.doman.tld", "subscriber")) {
# www_challenge("host.domain.tld", "0");
# exit;
# };
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
exit;
}
Any and all help appreciated.
--
Cliff Flood
Systems Administrator
Jazinga Inc.
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
I have a SIP provider: telecomitalia that use a cirpack!
The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...)
When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified.
Both TRYING/RINGING with session description Connection Address/port are not replaced!
OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address
because of force_rtp_proxy())
But other softphones like Twinkle don't understand it and still send RTP to the first IP!
Do you know how I can delete or modify TRYING/RINGING SDP?
Thanks
Below an example of a RINGING with SDP
#
U 212.129.6.65:5060 -> 88.191.45.91:5060
SIP/2.0 180 Ringing.
Allow: UPDATE,REFER.
Call-ID: 866489712(a)192.168.95.47.
Contact: <sip:212.129.6.65:5060>.
Content-Type: application/sdp.
CSeq: 21 INVITE.
From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903.
Record-Route: <sip:88.191.45.91;lr;ftag=542924903>.
Server: Cirpack/v4.39a (gw_sip).
To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67.
Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP
192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym.
Content-Length: 303.
.
v=0.
o=cp10 118890098142 118890098144 IN IP4 212.129.46.35.
s=SIP Call.
c=IN IP4 212.129.47.194.
t=0 0.
m=audio 32896 RTP/AVP 8.
b=AS:64.
a=rtpmap:8 PCMA/8000/1.
a=ptime:10.
a=sendrecv.
m=video 65534 RTP/AVP 34 31.
a=rtpmap:34 H263/90000/1.
a=fmtp:34 .
a=rtpmap:31 H261/90000/1.
a=fmtp:31 .
a=inactive.
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Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFG3TdoqjpLE0HiOBYRAjnkAKCgzB4p89oTKhxuvZcLlklTme9w3ACeP+TW
FI+LkwgbaJ1XJTIm3FUSMTg=
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Hi Guys
I upgraded my mediaproxy to 1.9.0 and found out sometime later that the
data usage update at call end seems to have stopped.
I'm using direct mysql (not radius) for the end of call updates.
No error message in the logs and a manual test of mysql show nothing wrong.
Any idea.
Mike
