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
Hello,
I have configured the SNMPstats module fine in my machine because I get the
SNMP information with the new KAMAILIO MIBS but I don't understand why the
value of kamailio dialogs is always 0:
KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0
KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0
KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0
KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0
However I can see the transactions:
KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32:
54797
Should I configure an specific parametr of the SNMPstats module?
Regards,
Hello everyone,
My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field.
Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core?
Thanks in advance. Best regards,
Bruno C.
Hi all,
I need to modify the host part of a contact header. I'm trying
something like:
route {
.....
if (remove_hf("Contact")){
if
(insert_hf("Contact: rn", "Contact"))
xlog("Contact modified");
}
if (!t_relay("YYY.YYY.YYY.YYY","5060")) {
sl_reply_error();
}
......
}
When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older
Contact field.
I try subst_hf with regexp, subst and simply remove
Contact, all this does not work.
What i am doing wrong?
--
С
уважением Зуев Михаил
For immediate release:
ATLANTA, GA (1 April 2015)--Evariste Systems LLC, an Atlanta-based software
vendor specialising in Kamailio-based service delivery solutions for the
VoIP ITSP market, is pleased to announce that it, in collaboration with
Red Hat Software and Ringfree Communications, has finalised the
absorption of the Kamailio SIP Server into the 'systemd' system management
platform for Linux. The new component shall be called 'systemd-rtc-server',
or 'Systemd Real-Time Communication Server'.
Alex Balashov, principal of Evariste and leader of the tri-vendor
collaboration effort, will officially announce the handover of the reigns
of the Kamailio project to the personal leadership of Lennart Poettering
at the upcoming Systemd Real Time Communications World conference, to be
held in Berlin on 27-29 May of this year.
John Knight, Director of GNOME 3 Integration and part-time usability
consultant at Ringfree Communications, based in Hendersonville, North
Carolina,was quick to summarise the triumphs of the long-standing
integration effort.
Remarked Knight:
"The industry has recognised for years that a SIP proxy is a basic building
block in the 'init' subsystem of any Linux host. In this age of multimedia
communication with voice and video, it was a travesty that systemd handled
time synchronisation, network configuration, login management, logging,
and console, but not SIP message routing."
Sean McCord, a veteran partner at Atlanta-based integrator CyCORE & Docker,
was quick to concur:
"SIP calls are much easier to troubleshoot with binary logs. Combined
with packet captures of TLS-encrypted WebRTC calls, systemd-journald
is the ultimate call setup troubleshooting methodology of the responsive,
kinetic enterprise."
To support the integration of Kamailio into the ecosystem of every major
Linux distribution, Evariste has released new 'dbus_api' and 'pulseaudio'
modules for the project.
Balashov stated, "We fully expect to use the D-Bus API to achieve
gnome-session integration with systemd-rtc-server-usrloc, but we aren't
going to leave Windows users behind; KamailioSvcHost.exe will support
Domain Controller policies for G.722 in Active Directory forests."
Despite an aggressive delivery timeline by the tri-vendor consortium behind
systemd-rtc-server, industry commentators have widely lambasted the fact
that it took so long for Kamailio to become integrated into systemd. Fred
Posner, solutions architect at The Palner Group in Fort Lauderdale, Florida,
recently wrote in a widely-publicised blog post:
"sr-dev have been keeping their heads in the sand for too long. For years
now, it has been completely obvious and self-evident to anyone with half
a brain that all kinds of VoIP software should be included in systemd.
It's a basic building block of the whole OS, having absorbed functionality
previously provided by all kinds of packages like util-linux and
wireless-tools."
John Knight of Ringfree accepted the criticism readily, but advocated a
forward-thinking orientation focused on breaking with the uncertainty of
the past:
"In the absence of a SIP component for routing calls to the PSTN, some
people thought, 'systemd has no clear direction apart from the whims of its
developers, and is a perpetually moving goal post.' Well, a SIP server
should
put an end to that whole discussion; that's exactly what was missing,
and now
that we have systemd-rtc-server, we've eliminated all doubts about the
coherence, conceptual integrity and finality of systemd."
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Good day,
I’m experiencing some problems with our VoiP providers handling of REGISTER requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 with SIP-Alg enabled. This setup kind of works with UDP but our provider wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done some wire tracing and to me it seems that the providers configuration is to blame, but then - there are many RFCs out there and many NAT and UAC bug workarounds. Anyway, I wanted to get the opinion of “the" experts about how the requests send to the UAS SHOULD be correctly interpreted.
The REGISTER requests/responses look like this (outside of the firewall):
Protocol TCP!
client port 19091 <-> server port 5060
REGISTER sip:pbx.peoplefone.ch SIP/2.0
Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec
From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485
To: "Michel" <sip:90780408050@pbx.peoplefone.ch>
Call-ID: 2825358480@10_10_128_10
CSeq: 1 REGISTER
Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>
Max-Forwards: 70
User-Agent: N720-DM-PRO/70.089.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec
From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485
To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc
Call-ID: 2825358480@10_10_128_10
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy"
Server: kamailio (3.2.1 (x86_64/linux))
Content-Length: 0
REGISTER sip:pbx.peoplefone.ch SIP/2.0
Via: SIP/2.0/TCP 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0
From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485
To: "Michel" <sip:90780408050@pbx.peoplefone.ch>
Call-ID: 2825358480@10_10_128_10
CSeq: 2 REGISTER
Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>
Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch", uri="sip:pbx.peoplefone.ch", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", response="764f371a08d258157a249f8d1b852514"
Max-Forwards: 70
User-Agent: N720-DM-PRO/70.089.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TCP 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0
From: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=2153084485
To: "Michel" <sip:90780408050@pbx.peoplefone.ch>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda
Call-ID: 2825358480@10_10_128_10
CSeq: 2 REGISTER
Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP"
Server: kamailio (3.2.1 (x86_64/linux))
Content-Length: 0
The ip:port the firewall is sending those requests from is ip 212.126.160.92 port 19091. So this does NOT match the port from the Contact header. For TCP this seems rather logical to me, as one cant be listening on a TCP port and use it for sending at the same time. The UAC closes this “register connection” with TCP FIN after the register, and so does the firewall.
However unfortunately subsequent requests from the provider (ie UAS) come in on port 19091 (not port 6717 from the Contact header) and the firewall simply drops them.
Observations:
- the server does NOT include received=212.126.160.92 in the Via of the reponse. According to RFC3581 this is mandatory when rport is present in the request, so this is probably an error in the server.
- the server does include received="sip:212.126.160.92:19091;transport=TCP” in the Contact of the response. I didnt see this in any RFC (which means nothing;-) but it could be an error.
- after the client received the 200 OK it closes the TCP connection.
- the server tries several times to re-contact the client (incoming TCP SYN). However not on port 6717 (defined in the Contact header) but on port 19091 (where the REGISTER came from).
RFC3581 defines special behaviour when “rport” is defined in the request (i.e. response should go to the same port the request came from) - however it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) requests from the server to the client. Those are strictly spoken not replies (or are they?).
RFC5626 defines that a “proxy” should keep track of the flows over which it received a registration and send requests over the same flow. It is not clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes an “ob” parameter in the Contact field if it whishes further requests over the same flow. Also the RFC mandates a client to add a "reg-id=x" in the Contact field. Both are not the case here, so in short I think RFC5626 should NOT be applied. In which case conecting to the originating port (instead of the Contact port) would be a server error.
So in short and if I interpret the RFCs correctly, the client is reachable and should be contacted on
Transport: TCP
IP: 212.126.160.92
Port: 6717
If anyone who lives and breathes SIP could enlighten me if the UAS is right to call back on 19091 instead of 6717 I would really appreciate it;-))
Best regards,
Joachim
Hi Daniel and others,
I'm having a problem with acc module if I'm using the event_route/
branch-failure:
say, the call comes from the app server and goes to the registered user.
We arm the the failure route and per-branch failure route for the 302
redirect from the UA. We explicitly reset the accounting flags because
we don't want to account the calls from the app server. The transaction
is created implicitly by the t_relay().
Now if the UA responds with the 302 response and we are going to process
that, we want to create an acc record for the new target from 302
message because this call may incur additional costs. I'm setting the
accounting flags and calling t_flush_flags() but that doesn't work (no
accounting record). Any idea if I'm doing something wrong or maybe
there's a bug when changing the flags and then calling t_flush_flags
from the event_route?
Here are the module parameters:
modparam("acc", "early_media", 0)
modparam("acc", "report_ack", 0)
modparam("acc", "report_cancels", 1)
modparam("acc", "detect_direction", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "db_url", PAIR_URL)
modparam("acc", "db_extra",
"src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")
modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)")
modparam("acc", "time_mode", 2)
modparam("acc", "time_attr", "time_hires")
modparam("acc", "cdr_log_enable", 0)
FTR, we tried 4.1.6 and 4.1.8.
And here is the event route (with flags defined like this:
flags FLAG_ACC_DB:1, FLAG_ACC_MISSED:2, FLAG_ACC_FAILED:3, ...):
event_route[tm:branch-failure:redirect]
{
route(ROUTE_STOP_RTPPROXY_BRANCH);
if($T_rpl($rs) == 301 || $T_rpl($rs) == 302)
{
# initialise variables when entering failure route
route(ROUTE_INITVARS);
# these need to be avps because we need it in reply/failure-route
$(avp(s:from_faxserver)[*]) = 0;
$(avp(s:to_faxserver)[*]) = 0;
$(avp(s:cf_from_pstn)[*]) = 0;
$(avp(s:from_pstn)[*]) = 0;
$(avp(s:proxylu_from_pstn)[*]) = 0;
$(avp(s:lcr_flags)[*]) = 0;
$(avp(s:em_call)[*]) = 0;
$(avp(s:from_pbx)[*]) = 0;
$(avp(s:p_to_device)[*]) = 0;
$(avp(s:p_to_group)[*]) = 0;
$(avp(s:is_primary)[*]) = 0;
# now let's process a 30x
$(avp(s:acc_state)[*]) = "cfb";
$(avp(s:orig_acc_caller_user)[*]) = $avp(s:acc_caller_user);
$(avp(s:orig_acc_caller_domain)[*]) = $avp(s:acc_caller_domain);
$(avp(s:acc_caller_user)[*]) = $avp(s:acc_callee_user);
$(avp(s:acc_caller_domain)[*]) = $avp(s:acc_callee_domain);
$(avp(s:caller_uuid)[*]) = $avp(s:callee_uuid);
$(avp(s:callee_uuid)[*]) = $null;
# the $var(no_acc) is 0 at this point but the flags may have
been reset
# if this is a call from PBX user - we do want accounting for
the 302 redirect
if(isflagset(FLAG_ACC_DB)) {
xlog("L_NOTICE", "++++++ ACC flag is set - [% logreq -%]\n");
} else {
xlog("L_NOTICE", "------ ACC flag is NOT set - [% logreq
-%]\n");
}
setflag(FLAG_ACC_FAILED);
setflag(FLAG_ACC_DB);
t_flush_flags();
# get last URI from destination-set and set it as R-URI
$var(contact) = $T_rpl($ct);
$var(contact) = $(var(contact){nameaddr.uri});
if($var(contact) == 0 || $var(contact) == $null)
{
xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302
- [% logreq -%]\n");
acc_db_request("480", "acc");
$var(announce_handle) = "callee_tmp_unavailable";
$var(announce_set) = $xavp(callee_real_prefs[0]=>sound_set);
$(avp(s:announce_code)[*]) = 480;
$(avp(s:announce_reason)[*]) = "Temporarily Unavailable";
route(ROUTE_EARLY_REJECT);
}
$ru = $var(contact);
xlog("L_NOTICE", "Redirect from UAC intercepted - [% logreq -%]\n");
$(avp(s:forwarder_cli_userprov)[*]) = $T_rpl($tU);
$(avp(s:forwarder_domain_userprov)[*]) = $T_rpl($td);
$var(forward) = 1;
$var(redirected_forward) = 1;
route(ROUTE_LOAD_CALLER_PREF);
route(ROUTE_FIND_CALLEE);
}
}
Thanks.
Andrew
Hello all,
Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio.
Especially creating partitions, DRBD devices and mount points.
Thanks in advance,
-Sid
"May the light be with you." ______________________________________________
Siddhardha Garige
www.luminepixels.com
