Hello,
Kamailio SIP Server v4.2.6 stable release is out.
This is a maintenance release of the previous stable branch, 4.2, that
includes fixes since release of v4.2.5. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.2.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.2.6 (or to 4.3.x series).
For more details about version 4.2.6 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2015/07/kamailio-v4-2-6-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the latest stable branch is 4.3, at this moment with its latest
release v4.3.1. See more details about it at:
* http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/miconda - http://www.linkedin.com/in/miconda
Looking for some doc that describes how the new columns in 4.3.x location
table is supposed to be used. Can find some email discussions about this,
but seems no formal doc exists. Would appreciate if someone can provide an
explanation.
Thanks
(www.)kamailio.org is unreachable at this time (the whole morning) on
2a00:d60:0:400::2. ipv4 is responsing.
Never noticed whether connectivity via ipv6 was working in the past or this is
new and misconfigured.
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/
Greetings.
Sorry for previous false letter.
I want to check if user is registered before reply is routed to its
target user agent.
Imho it should be like this:
onreply_route[MANAGE_REPLY] {
if ($rm=="INVITE") {
if (registered("location", $ru)) {
# Do job - pause for 15 seconds and give chance to reregister
}
}
}
But kamailio complains on registered function
0(6274) ERROR: <core> [cfg.y:3295]: yyparse(): misused command registered
0(6274) : <core> [cfg.y:3439]: yyerror_at(): parse error in config
file /usr/local/etc/kamailio/kamailio.cfg, line 1121, column 35:
Command cannot be used in the block
Documentation confirms - registered() function can be used in
REQUEST_ROUTE and FAILURE_ROUTE only.
Is it possible to check if user is registered when reply is processed
in scripts?
Thank you!
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
I have Kamailio in a local network with RTPPROXY and NAT MANAGE
I have seen an issue with ACK as show when Asterisk sends ACK to Kamailio,
it sends to advertised Address instead to Kamailio IP address.
What could i add to solve this?
route[WITHINDLG] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if (is_method("BYE")) {
setflag(FLT_ACC); # do accounting ...
setflag(FLT_ACCFAILED); # ... even if the
transaction fails
dlg_manage();
}
if ( is_method("ACK") ) {
# ACK is forwarded statelessy
route(NATMANAGE);
}
route(RELAY);
} else {
if (is_method("SUBSCRIBE") && uri == myself) {
# in-dialog subscribe requests
route(PRESENCE);
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction
... ignore and discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
}
BR
Hello,
I am planning to release v4.2.6 later this week, most likely on
Thursday, July 30.
If there is anything not reported that should be considered for this
minor release of the branch 4.2, write to sr-dev mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
Hello everybody,
I am currently using Kamailio in order to separate traffic based on dialed
number.
Some traffic goes to several Asterisk in a load balance strategy and other
traffic goes to other SIP devices.
I use carrierroute module in order to route traffic based on destination. I
use scan_prefix to classify traffic and rewrite_host to route the traffic.
A destination number usually go to one of 4 asterisk server. I do this in
order to load balance the traffic. In order to distribute traffic I use
prob (0.25 for each destination).
Using this configuration, I have to include on carrieroute table a row for
each destination node. Since the amount of destination is huge, I have to
repeat configuration in tables. It also complicates maintenance.
I was thinking of using a combination of carrierroute and dispatcher
module. Carrierroute for classify traffic, strip numbers, and rewrite
destination host and dispatcher to distribute the traffic over asterisk.
Using this configuration I have to indicate asterisk nodes once in
dispatcher table once. Also in carrierroute table I have to indicate each
prefix once.
I was thinking in something like this.
if(!cr_route("carrier", "domain", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not allowed");
drop;
}
if(!ds_select_dst("1", "4")) {
sl_send_reply("403", "Not allowed");
drop;
}
t_relay;
Does it make sense, or am I complicating everything and there is a magic
way to achieve this.
Thank you.
