Hi,
Can I re-route a SIP response message? In other words, after the SIP proxy
receives a 503 from node A, I need to change the TO field and re-route it
to node B. I tried to modify
$rd or $rU and didn't work.
Thanks,
R
Hello,
a new module is now on development branch, named msrp, offering an
embedded MSRP relay. Message Session Relay Protocol is specified by IETF
( core and relay part in RFC4975 and RFC4976), one of typical use cases
is Instant Messaging sessions negotiated via INVITE-200ok-ACK, a
different approach than SIP MESSAGE request.
There is no external dependency, the transport layer from core is reused
to manage TCP/TLS connections used for MSRP. A MSRP relay is pretty much
an independent node, thus you can run Kamailio just to deal with MSRP
traffic. But there is no problem to run Kamailio to handle SIP and MSRP
traffic in the same instance, same or different sockets (it is working
even when SIP/MSRP are sent over the same connection or different
connections to same port).
I couldn't find a reliable and trustable open source SIP phone to test
with it so far (hints are welcome), the module was tested with network
tools and chaining Kamailio instances. Help with testing and feedback is
very appreciated -- I can provide guidelines to adjust the config file
to fit tester's needs, just contact me off list or via IRC channel.
For the moment, the relation (user,session) management is done in the
config, using htable module for example (see the README), it may be
added inside the module for the future to make it easier overall,
although is less than 10 lines of config -- this is mainly for extra
security reasons, to check if the session id matches the connection that
was opened when the session was created, otherwise the user
authentication functionality is done reusing existing functions from
auth module. Another benefit is performances, storing local socket and
connection IP/port saves some time to lookup the connection.
Building MSRP relay on top of SIP server was done first for the benefit
of reusing the transport layer from the core for IPv4/IPv6 and TCP/TLS,
which is mature, scalable and offers asynchronous communication. Besides
that, you have most of the config tools to route SIP requests available
for routing MSRP (authentication, authorization, IP checking,
accounting, a.s.o.).
I made a news post with more details:
* http://www.kamailio.org/w/2012/01/new-module-embedded-msrp-relay/
The readme of the new module is available at:
* http://kamailio.org/docs/modules/devel/modules/msrp.html
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hello everyone,
as has become already a tradition, we would like to invite all VoIP
people and everyone interested in sip-router and related projects
(SER, Kamailio/OpenSER, OpenSIPS, SEMS etc) who is attending FOSDEM,
Europe's biggest Free and Open Source conference, to a dinner on the
evening of Saturday February 4th. This is the perfect opportunity to
socialize among the developers, get to know the people behind email
addresses and commits, do networking, find the people for the next
project, and/or just hang out together with fine food and drinks.
Participation at the event is free of charge, everybody pays for their
own expenses. In order to do reservation in a restaurant please send
me a short mail if you would like to attend. Please also indicate if
you don't want to have your contact details communicated to the other
social event attendees.
If you are still wondering whether it's worth to go to FOSDEM - among
all the other interesting talks this year in the telephony-devroom on
Sunday, 5th, there will be a few directly related talks (the exact
time may be subject to change):
11:30 Andreas Granig: From zero to VoIP provider in 15 minutes
12:55 Stefan Sayer: Session Border Control with SEMS
15:25 Daniel-Constantin Mierla: Secure SIP Communications with Kamailio
16:15 Saúl Ibarra Corretgé: SIP beyond VoIP
http://fosdem.org/2012/schedule/track/telephony_and_communications_devroom
Hope to see you all in Brussels!
With Best Regards
Stefan Sayer
--
tel:+491621366449
sip:sayer@iptel.org
mailto/xmpp:stefan.sayer@gmail.com
Hello all,
I'm happy to announce a new developer for the project: Anca Vamanu.
Welcome to the development team!
Anca joined 1&1 last year and worked since then on several internal modules
and also many important bug fixes for our kamailio server setup. She is a
experienced C and C++ developer and contributed in the past a lot to OpenSER
and other VoIP projects.
One interesting work she just finished and that we also like to contribute to
the community is a db_cassandra module, providing access to this NoSQL
database [1] from kamailio. She will take over the maintenance of this module
and also work on improving other parts of the server.
Best regards,
Henning
[1] http://cassandra.apache.org/
--
Henning Westerholt - Head of IT Operations Internet Access & Communications
1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany
Hi All,
I spent a few hours this afternoon setting up a Homer SIP Capture
server today. The build went well, good example setup howto's. My
setup is a single kamailio/sipcature/homer node sitting on a LAN with
2 NIC's, eth0 for network access ssh/web, eth1 is the capture port
connected to switch mirrored port. On the switch I have a kamailio
proxy server being monitored.
All SIP traffic through the proxy server is logged to the homer
server, but the homer server is receiving SIP packets on the capture
interface and sending out SIP messages through the network access eth0
port. The messages are being processed from the on_reply route. The
default config for the capture node is to capture all SIP packets, but
this seems to create the problem. Here is the route and onreply_route
default config:
route {
sip_capture();
exit;
}
onreply_route {
sip_capture();
exit;
}
If I replace the onreply_route exit; with drop; then the capture
server no longer sends SIP messages out the eth0 interface and
everything seems to work as expected.
So my question; what is the difference between a drop; and exit;
within the on_reply route and how the hell is the capture server
sending messages out the eth0 network interface when the server is
100% bound to eth1 capture interface?
Any insight or guidance is appreciated.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
I was wondering if any NoSQL adapters are available for Kamailio to do
"subscriber authentication" backend?
I was looking at Mongo or CouchDB as two possible backends
Seems like these would be good candidates for Backends for Kamailio.
Any ideas on this?
Thanks
David.
Hello all,
We are currently in the process of reworking our billing platform which uses Kamailio for routing in conjunction with a Django based web interface. I am curious if anyone has any feedback on which database engine they chose for their deployment and why. Obviously performance is a concern but data integrity is essential. Currently we have two Kamailio based clusters in two datacenters using MySQL in a dual master setup for each of the primary database nodes. I've been looking into PostgreSQL as a possible alternative and would like to hear any community thoughts.
Thanks,
Spencer
Hello Yaron
can you try "mi mt_reload"?
Regards
Javi
>
> Message: 2
> Date: Mon, 16 Jan 2012 11:54:04 +0200
> From: Yaron Nachum <nachum.yaron(a)gmail.com>
> Subject: [SR-Users] using XMLRPC module for fifo commands
> To: sr-users(a)lists.sip-router.org
> Message-ID:
> <CADKX-JDqsxBb3T6XOsiUUs=Xj+idNAXEECBx1HeuAu8dSEdbsA(a)mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi everyone,
> Is it possible to use the xmlrpc module in order to send fifo commands to
> the kamailio. for example I would like to use it in order to trigger mtree
> table reload.
>
> I have already enabled the xmlrpc module and it responds. however I didn't
> succeeded in sending the appropriate syntax in lrder to make the mtree
> reload work.
>
> I tried using mtree.mt_reload method or fifo.mt_reload .
>
> Does anyone have a clue?
>
> Yaron.
>
Hi everyone,
Is it possible to use the xmlrpc module in order to send fifo commands to
the kamailio. for example I would like to use it in order to trigger mtree
table reload.
I have already enabled the xmlrpc module and it responds. however I didn't
succeeded in sending the appropriate syntax in lrder to make the mtree
reload work.
I tried using mtree.mt_reload method or fifo.mt_reload .
Does anyone have a clue?
Yaron.
Hi,
this is my cfg:
if(do_routing("0")){
$avp(tdr)=$avp(dr_attrs);
} else {
$avp(tdr)="1";
}
subst_user('/(.*)/$avp(Ssrvindx)/');
xlog("L_CRIT","$C(rg) group is $avp(tdr)$C(xx)\n");
if(!do_routing("$avp(tdr)")){
xlog("L_CRIT","$C(rg)No TDR is found for $avp(DID) with Index
$avp(Ssrvindx)$C(xx)\n");
t_reply("404", "Not found");
exit;
}
in the log, i get the following:
1(25799) CRITICAL: <script>: group is 8
1(25799) ERROR: drouting [drouting.c:706]: failed to get group id
when i change the do_routing("$avp(tdr)") to do_routing("8"), everything
works.....
it looks like with the avp (although it is 8) it does not work.
any ideas?
