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've a very strange scenario to work on which requires me to parallel fork
the call to the same Destination provider. The only problem here is that
they think that the second INVITE with different branch tag is a
re-transmission and hence only take one call forward.
I do not have to modify any R-URI or any headers, hence just
'append_branch() before t_relay() is in the code.
...
$ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port);
append_branch();
route(RELAY);
...
*Question:* Is there anyway possible I can change the CallID of the forked
INVITEs ?
I have tried using Topoh module, but it still puts the same CallID before
sending out. Different from the A-leg but the Sent out INVITEs have same
Call-ID value.
In other weird scenario, I've also tried branching, and looping call within
Kamailio before sending out to carrier hoping that Kamailio would treat the
two different calls and Topoh would change the Call-ID on both INVITEs
before sending out..
...
if(is_present_hf("X-FORKED")) {
$ru = "sip:" + $rU + "@" + $avp(carrier_ip) + ":" + $avp(carrier_port);
route(RELAY);
}else {
append_hf("X-FORKED: 1\r\n");
append_branch();
route(RELAY);
}
...
Still I get same Call-ID on outgoing branched call.
I could branch out one INVITE to a MediaServer, say FreeSWITCH/Asterisk but
again I don't want to have that component bottleneck the throughput. That
could be my very last option.
Looking for some ideas.
Regards,
Sammy
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
Hi Guys,
Im using a keepalived solution between 2 kamailio boxes, with
net.ipv4.ip_nonlocal_bind
=1 .
On the "non master" I pretty much constantly get these logs in syslog.
What Im looking for is a way to suppress these because they are expected in
my configuration.
is there any way to tell kamailio to not log this... ( or at least, not at
CRIT and ERROR )
*Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: tm
[../../forward.h:208]: msg_send(): udp_send failed*
*Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: ERROR: <core>
[udp_server.c:576]: udp_send():
sendto(sock,0x7fc0732d5ab0,380,0,xxx.xxx.xxx.xxx:5060,16): Invalid
argument(22)*
*Jun 25 10:32:53 lb2 /usr/sbin/kamailio[24869]: CRITICAL: <core>
[udp_server.c:581]: udp_send(): invalid sendtoparameters#012one possible
reason is the server is bound to localhost and#012attempts to send to the
net*
--
Sincerely
Jay
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 there,
I'm facing some issues trying to install siremis with postgresql.
The file Config.xml has the correct configuration for postgresql driver but
It seems that during installation phase siremis can't create database
structure for postgresql.
Anyone have installed siremis with postgresql, if yes can you share the
steps needed for this installation.
--
Cumprimentos
José Seabra
How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server?
Great regards,
Victor Olvera
Hi,
Trying kamailio with Ubuntu 16.04 and I'm getting errors as follows:
ERROR: tls [tls_init.c:490]: tls_pre_init(): Unable to set the memory
allocation functions
The error is identical to the one mentioned here:
https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992
Is there a solution to this. Thanks for any response.
- Jayesh
