Hi,
I'm trying to set up a Kamailio (4.1.3) server which converts between IPv6
and IPv4. Everything looks pretty good. Now I have a test client which
sends packets via IPv6, but in contact header, SDP etc. there are still
IPv4 addresses. (This comes from some converting from v4 to v6 earlier.)
Thus, Kamailio tries to do a fix_nated_contact(). But after fixing, the
Contact URI is wrong.
Example:
inbound
Contact: <sip:bob@192.168.8.132>
outbound
Contact: <sip:bob@1234:1234:0:1234:0:0:0:2>
The correct way of transforming the Contact URI would be:
Contact: <sip:bob@[1234:1234:0:1234:0:0:0:2]>
This incorrect Contact URI doesn't hurt first, but when the called party
wants to hang up the call, it addresses the URI from Contact header, and
Kamailio can't parse the Request URI:
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: pv
[pv_core.c:304]: pv_get_ruri_attr(): failed to parse the R-URI
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: rr [loose.c:934]:
loose_route(): failed to parse Request URI
Jan 22 12:03:48 router /usr/sbin/kamailio[21309]: ERROR: domain
[domain.c:140]: is_uri_host_local(): error while parsing R-URI
Is this a bug in the nathelper module? Was it never meant to handle IPv6
addresses? Or do I understand something wrong?
Best Regards,
Sebastian
Hi All,
I have a strange issue, I have set the module parameter for nathelper's
force_socket to a specfic ip/port, however, when I perform a sip trace I
can see that all locally generated options messages are not sent from
the socket defined in the modules parameters.
I am not setting $fs anywhere in the script, so I am not over-writing
it. Do any of the other module parameters have a bearing on if nathelper
uses the force_socket parameter?
My current nathelper settings are:
modparam("nathelper", "received_avp", "$avp(RECEIVED)")
modparam("nathelper", "natping_interval", 20)
modparam("nathelper", "natping_processes", 4)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_from", "sip:keepalive@domain.com")
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "sipping_bflag", NAT_BFLAG)
modparam("nathelper", "force_socket", "1.2.3.4:5060")
I am using kamailio v4.3.1:
# kamailio -V
version: kamailio 4.3.1 (x86_64/linux) f38e67
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST,
HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: f38e67
compiled on 18:15:23 Jul 20 2015 with gcc 4.4.7
Thanks
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 am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I
have add and registered two IP phones with kamailio these phone able to
call each other.
How can i make outbound call with that I am using a asterisk server as PSTN
Gateway. i am not famalier with kamailio script.
Regards
Sharad Tyagi
SparkTG info Pvt Ltd
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
Hi fellow kamailio users,
I'm playing actually with t_on_branch_failure, in a simultaneous forking
scenario ( lookup("location") returns multiple contacts ).
Any idea on how to get those "on_branch_failure" working ?
Thanks in Advance !
* Test by putting failure_exec_mode:
- at 1 => no branch failure being called, never.
- at 0 => t_relay does not fail if a single branch answers the
transaction, even if all other branches are failing.
* Test by moving the t_on_branch_failure("manage_branch_failure"); line
before or after the lookup("location") or on a _on_branch route to set
it per branch.
- The on_route branch are executed, but never the failure one.
- tried also with failure_exec_mode = 0 or 1.
* Using set_forward_no_connect, I can see in the logs a "Sending to
branch X failed" but no failure route being executed. Without it,
nothing in the logs indicating that the branch has timeout.
* Config extract:
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3) # 3 by default
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
modparam("tm", "reparse_on_dns_failover", 0)
# 1: Allow branches to have a per branch failure route
# 0: t_relay should fail if a branch fail
modparam("tm", "failure_exec_mode", 1)
# Allow locally generated 408 to be forgotten in case a reply is received.
#modparam("tm", "faked_reply_prio", 2000)
# USER location service
route[LOCATION] {
t_on_branch_failure("manage_branch_failure");
if (!lookup("location")) {
....
}
# Let's get back after 2.5s if no 100/Trying received.
t_set_fr(0, 2500);
# set_forward_no_connect();
#t_on_branch("SET_BRANCH_FAILURE");
route(RELAY);
exit;
}
route[RELAY] {
....
if (!t_relay()) {
xlog("L_WARN","$rm|$ci|$fu|$tu|RELAY_FAILED");
sl_reply_error();
}
exit();
}
# set failure route per branch
branch_route[SET_BRANCH_FAILURE] {
xlog("L_WARN","$rm|$ci|$fu|$tu|SETTING_BrANCH_FAILURE_ROUTE");
t_on_branch_failure("manage_branch_failure");
}
# manage failure routing cases per branch
event_route[tm:branch-failure:manage_branch_failure] {
xlog("L_WARN","$rm|$ci|$fu|$tu[$avp(dest_to)]|MANAGE_BRANCH_FAILURE");
}
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
