Hi,
I have two SIP servers, *A* and *B*, connected each other though a OPENVPN
tunnel. The server *B* needs to t_relay() every SIP message containing the
method MESSAGE to the server *A* but these messages never reach destination.
I have tested the tunnel connectivity and works fine. I wrote a Perl script
(located in *B*) that sends SIP MESSAGES to Kamailio (located in *A*) trying
to figure out what is happening but these messages are received by *A* and
processed correctly but when *B* does the same from Kamailio, it is never
received.
Here is the route part of kamailio.cfg in *B*:
Observation: ($rU == "1004") result is *true*
*
if(is_method("MESSAGE"))
{
if($rU == "1004")
{
xlog("L_INFO","En 1004");
rewritehost("10.8.0.1");
if (!t_relay())
xlog("L_INFO","MIO Error en t_relay");
t_reply("200", "Ok");
xlog("L_INFO","MIO despues rewrite");
exit;
}
.....*
The perl script that WORKS:
*$msg = 'MESSAGE sip:1004@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2
From: "2002" <sip:2002@192.168.1.2 <sip%3A2002(a)192.168.1.2>>;tag=1837944796
To: <sip:1004@192.168.1.2 <sip%3A1004(a)192.168.1.2>>
Call-ID: 19722852989(a)192.168.1.2
CSeq: 15773 MESSAGE
Contact: <sip:2002@**PublicIP-protected**:5060>
Max-Forwards: 29
User-Agent: DBL
Content-Type: text/plain
Content-Length: 34
+595981[protected]
hello from kamailio
';
use IO::Socket;
my $sock = IO::Socket::INET->new( Proto=>'udp',
PeerHost=>'192.168.2.102',
PeerPort=>'5060');
print "Sending msg $msg\n";
$sock->send($msg) or die "error sending $!\n";
*Please help!
Thanks in advance.
Carlos.
Hi,
About 3 weeks ago i upgraded one of my production server with latest stable
kamailio version 4.2.1-fad00a. Now i am getting a lot of complaints about
missing CDR events in ACC table. I observe following problems,
1. There are only BYE records in acc table, no record for INVITE or ACK.
2. In kamailio logs when ACK is received against 200 OK response for
INVITE, i see following errors,
--
ERROR: <core> [parser/parse_from.c:113]: parse_from_uri(): failed to parse
>From uri
ERROR: pv [pv_core.c:434]: pv_get_xto_attr(): cannot parse From URI
NOTICE: <script>: [udp:<null>@1.0.0.127:5060]: Call from 'you(a)kamailio.org'
to 'you(a)kamailio.org' has been hanged up by '<null>' at '1419364717.255484'
--
Of course all these errors are bogus, I have checked all headers in ACK
(not just FROM header), they all seem perfectly fine and valid.
3. Then the dialog times out,
--
WARNING: dialog [dlg_handlers.c:1440]: dlg_ontimeout(): timeout for dlg
with CallID '6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9' and tags
'1D3ECD34F5731AB845BA3064AC95BB2D'
'7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4'
--
4. Any further sequential requests complain about "unable to find dialog",
e.g.
--
NOTICE: <script>: Sequencial 'BYE' request received from caller
ERROR: uac [replace.c:591]: restore_uri(): new URI [] shorter than old URI [
sip:00xxxxxxxxxx@sip.domain.com]
WARNING: dialog [dlg_handlers.c:1174]: dlg_onroute(): unable to find dialog
for BYE with route param '5ae1.d595' [7845:22877]
--
5. However the acc record for BYE is written to db and log file,
--
NOTICE: acc [acc.c:318]: acc_log_request(): ACC: transaction answered:
timestamp=1419364760;method=BYE;from_tag=7f55e81e0630-100007f-13c4-6009-2440a4-5fa31570-2440a4;to_tag=1D3ECD34F5731AB845BA3064AC95BB2D;call_id=6D8BD23CAC65AE3C1DE1D0B531F87B8CFEAA9CB9;code=200;reason=OK;src_user=00xxxxxxxxxx;src_domain=
sip.domain.com
;src_ip=xx.xx.xx.xx;dst_ouser=+1xxxxxxxxxx;dst_user=1xxxxxxxxxx;dst_domain=yy.yy.yy.yy
--
The same config was working fine with older version 4.2.0-97cab8. The
kamailio config i am using is pretty much standard,
--
#!define FLT_ACC 1
#!define FLT_ACCMISSED 2
#!define FLT_ACCFAILED 3
#!define FLT_DLG 4
...
modparam("acc", "early_media", 1)
modparam("acc", "report_ack", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 1)
modparam("acc", "log_flag", FLT_ACC)
modparam("acc", "log_missed_flag", FLT_ACCMISSED)
modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)
# log to db
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db_missed_flag", FLT_ACCMISSED)
modparam("acc", "db_url", "DBURL")
...
request_route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
# handle requests within SIP dialogs
route(WITHINDLG);
# CANCEL processing
if (is_method("CANCEL")) {
if (t_check_trans()) {
t_relay();
};
exit;
};
#### only initial requests (no To tag) ####
t_check_trans();
....
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_DLG); # create dialog
setflag(FLT_ACC); # do accounting
setflag(FLT_ACCFAILED); # ... even if the transaction fails
$avp(dlg_timeout) = 60;
dlg_manage();
....
}
--
Any ideas why its happening? Since it is 3 weeks old so may be problem has
already been spotted and fixed by someone else. Otherwise let me know how
can i provide more info to help fix this issue.
Thank you.
Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP.
I read that STUN is for non-symmetric NAT, and RTP server is for symmetric
NAT.
Supposed A calls B.
If A, B both use symmetric NAT and STUN, they cannot hear each other
If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each
other.
Why is that?
I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt
for Prediction Failure, is that related to this problem ?
--
Khoa Pham
HCMC University of Science
Faculty of Information Technology
I'm involved in setting up a Kamailio instance and was hoping to make use of the auth_ephemeral module for authentication. But the module documentation doesn't quite make clear how exactly the module is to be employed or the web service set up. Does anyone have a working example?
Regards,
Hemanshu Vadehra
hemanshu.v(a)directi.com
Hi all,
I'm switching my development environment from Linux to Mac.
Is there any tutorial on how to setup Kamailio on this OS? It may be an
Unix flavor, but it's still quite different from Linux :).
Thanks,
--
Carlos
http://caruizdiaz.comhttp://ngvoice.com
+52 55 3048 3303
Hello All,
I have a need to save the Record-Route header from the INVITE and
use it as a route header in requests like BYE.
PROXY --> Kamailio(as PROXY) --> Asterisk
I am using rr module to add a record_route when passing the INVITE to Asterisk.
Thanks,
Jignesh Gandhi
Im trying to enable Kamailio TLS but the configuration is not running at
all when I cahnge it liike this as follow.
Kamailio 3.3 ubuntu 12.04 LTS
#!define WITH_TLS
#!ifdef WITH_TLS
enable_tls=yes
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/etc/kamailio/tls.cfg")
#!endif
Please help
--
Kethzer Docteur
Hi,
I would like to append a header to a 'winning' negative reply in
failure_route and let the Kamailio do the default action (state fully
forward the winning reply).
When I use append_to_reply("Foo: bar\r\n"); and then call exit; in
failure_route nothing is appended.
When I use same append_to_reply then t_relay("505","Error"); and exit; the
header is appended.
When I use append and t_reply with dialog modul turned on I got a bug I'm
solving here '[SR-Users] t_reply in failure route with dialog module'.
So my question is how to put a header into a reply when I don't want to
alter its code or text?
I'm using Kamilio 4.0.4 on Debian 7.1
Thanks for an answer
Efelin
Hi All,
I have build Kamailio 4.0.1 from source in CentOS 5.8(i386
architecture).I followed all instructions from
http://www.kamailio.org/wiki/install/4.0.x/git.(Though Modules_k directory
is not generated).I then edit kamailio.config file for websocket support as
described in webocket.cfg file in exmples directory.But while starting
kamailio it gives following error
ERROR: load_module: could not open module : libunistring.so.0: cannot
open shared object file: No such file or directory
0(30270) : [cfg.y:3567]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 318, column 12-57: failed to
load module
I checked websocket.so file in the specified directory and it is
already there.Can you please help me what's wrong with it?
please help.
Rupayan Dutta
Hello,
First, i have 7 years experience with Asterisk, but I started a project
with Kamailio, forgive me in advance if I say silly things...! ;-)
I set up a classic Asterisk / Kamailio configuration:
sip phones -> kamailio -> asterisk -> sip trunks/pstn.
When a call comes from the PSTN side, if I configure Asterisk as follows:
[012345678]
type = friend
username = 012345678
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com
fromuser = 012345678
Standard mode:
exten => 012345678, 1, Dial(SIP/012345678) -> The call is redirected on the
phone by Kamailio ! :-)
------------------------------------------------------------------------------------------------------------------------------------------------
Trunk mode:
[mytrunk]
type = friend
username = mytrunkUser
secret = xxxxxxx
host = dynamic
fromdomain = sip.mydomain.com
fromuser = mytrunkUser
exten => 012345678, 1, Dial(SIP/mytrunk/012345678) -> The call is rejected
by Kamailio....
exten => 012345679, 1, Dial(SIP/mytrunk/012345679) -> The call is rejected
by Kamailio ....
My question is how to allow the routing of multiple numbers (trunk mode) in
a SIP account with Kamailio?
Best regards,
Mickael
