Hi,
I'm having a problem with routing of BYEs in my multi homed Kamailio.
My setup is a Kamailio v4.1.5 with two NIC (172.20.11.0 & 192.168.10.0) and three phones. Two phones (1 for each LAN) are in a group, so using alias_db I can call them in parallel mode. Caller is on 172.20 network...When I answer from phone on 192.168.10.0, RTP is ok ... but on BYE Kamailio replies with 404 Not Here...and the RURI of BYE has the IP of Kamailio instead the IP of the callee.When I answer from phone on 172.20.11.0, RTP is still ok and also BYE is managed correctly from Kamailio and the RURI of BYE has the IP of the callee.
How can i solve this issue?Many thanks
Hi Spencer,
We use a similar approach as yours and use 302 Redirects coming from
Kamailio instead of proxying the call (in our case we do it via LCR
module in Kamailio). In FreeSWITCH you can then manually route these
redirects and get your CDR data out of it.
Cheers,
DanB
On 22.09.2014 12:00, sr-users-request(a)lists.sip-router.org wrote:
> On 18/09/14 17:34, Spencer Thomason wrote:
>> >Hello all,
>> >I have a need to determine which calls go to which gateway. The challenge is that CDRs are produced by a FreeSWITCH instance upstream of a provider side Kamailio proxy that handles route selection via drouting module. The topology is like this:
>> >
>> >customer proxy -> FS [1-N] (accounting) -> carrier proxy (route selection)
>> >
>> >Does anyone have any ideas how to make the FreeSWITCH instances aware of which gateway actually received the call?
Hi all,
I am using Kamailio to front my freeswitch servers and it is working great.
I am getting some problems with authentication, sometimes users which are
not registered with kamailio are also able to make calls. I have a feeling
that I am doing something really silly with kamailio routing. Following is
the AUTH route that I am using, please let me know I am missing something
very obvious.
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_FREESWITCH
if(route(FSINBOUND))
return;
#!endif
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$fd", "0");
exit;
}
# user authenticated - remove auth header
if(!is_method("REGISTER|PUBLISH"))
consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
sl_send_reply("403","Not relaying");
exit;
}
#!endif
return;
}
route[FSINBOUND] {
if($si== $sel(cfg_get.freeswitch.bindip) &&
$sp==$sel(cfg_get.freeswitch.bindport))
return 1;
return -1;
}
I have set "WITH_AUTH" and "WITH_FREESWITCH". Let me know, if any more
details are required.
System configuration:
Ubuntu: 12.04 LTS
Kamailio: 4.1.5 (x86_64/linux)
Freeswitch: 1.4.7 stable release
--
Thanks and Regards,
Ashwin Jain
Hello all,
I have a need to determine which calls go to which gateway. The challenge is that CDRs are produced by a FreeSWITCH instance upstream of a provider side Kamailio proxy that handles route selection via drouting module. The topology is like this:
customer proxy -> FS [1-N] (accounting) -> carrier proxy (route selection)
Does anyone have any ideas how to make the FreeSWITCH instances aware of which gateway actually received the call?
Thanks!
Spencer
Hello,
The port has just been committed to the FreeBSD ports tree and has been
mapped to:
/usr/ports/net/kamailio
It can be compiled and installed with FreeBSD's traditional:
# make install clean
It compiles on 8.x, 9.x and 10.x for the i386 and x86-64 architectures.
Possibly for the others archs too, but they to my knowing haven't been
thoroughly tested so far. We'll find out on the way. The way things look, I
will be the ports defacto maintainer.
After finishing my first draft of the port about a year ago, I ran a few
tests between a Kamailio server on FreeBSD and a Windows host using
MySQL as a phonebook and Jitsi as a frontend GUI. From the Kamailio point
of view, things worked fine, however, the only problem was, that I wasn't
really able to test voice functionality, since jitsi had some problems
accessing soundcard features on FreeBSD. Maybe things have changed since
then and/or this will give rise to new research in this area.
In case you are interested and for your cross-reference, you can find the
entire discussion that took place concerning the port here:
https://bugs.freebsd.org/bugzilla/show_bug.cgi?id=181301
Moreover, to this e-mail attached, you will find a shell archive
(kamailio.shar) of the port the way it has been mapped into the ports tree.
Enjoy using Kamailio on FreeBSD!
Oliver Mahmoudi
On Tue, Jul 23, 2013 at 7:51 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
> wrote:
> Hello,
>
>
> On 7/23/13 1:31 AM, Oliver Mahmoudi wrote:
>
>> Hello,
>>
>> I am currently working on porting Kamailio 4.0.2 to the
>> FreeBSD ports collection: /usr/ports, and I just have a
>> few general questions regarding compilation with regard
>> to the extra modules.
>>
>> There are 172 modules that come with Kamailio, and most,
>> however not all of them are compiled by default:
>>
>> # gmake all
>>
>> The base server and the base modules compile fine on
>> my 9.1 release.
>>
>> To really compile all of the 172 modules, Kamailio_4_0_2/INSTALL
>> says that the following external libraries (LIB_DEPENDS I assume)
>> are needed:
>>
>> /usr/ports/databases/mysql-client-*
>> /usr/ports/databases/postgresql-libpqxx
>> /usr/ports/textproc/expat2
>> /usr/ports/textproc/libxml2
>> /usr/ports/net/radiusclient
>>
>> This is put down as information for FreeBSD 5.4. Does that still hold
>> current for 9.1?
>>
>> Is there an exhaustive list of all the modules that the all target
>> does/does not compile and - apart from mysql, postgresql - which of the
>> extra modules require which libraries?
>>
>> I would set compile time options accordingly.
>>
>> This would greatly facilitate my work!
>>
> one file to look in is Makefile.groups, there you will see lists with
> modules organized by common dependencies. In debian, the packages are
> generated for each module_group_k... Each such group includes a list of
> modules defined in the same file.
>
> There are some comments giving hints about dependecies, but a better place
> is to look in the README of the module, where is a dedicated section for
> such details.
>
> We had some ports done long time ago, you find them in sources
> pkg/kamailio/.If you get up to date version, we can replace the old one in
> our repository as well.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
>
Hi to all,
is it possible to define same alias for differents SIP ID? How?
Documentations of "Alias_DB Module" says:
"If the alias resolves to many SIP IDs, the first is replacing the R-URI, the rest are added as branches"
..but i've tried to add more SIPID with the same alias but system doesn't allow it.
A.
I've installed Kamailio on multi NIC server (192.168.1.0 & 172.16.1.0), and I've configured rtpproxy in bridged mode in order to manage RTP but I've some trouble during call between endpoint placed on different network.
I've find an example regarding Kamailio+RTP in IPv4+IPv6..it's a good point to start but i can't use "af" function... I've tried to replace it with $si but only some call works.
How can i replace af correctly? Is there any example conf for my scenario?
Many thanks
Hello,
Kamailio SIP Server v4.1.6 stable release is out.
This is a maintenance release of the latest stable branch, 4.1, that
includes fixes since release of v4.1.5. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.1.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.1.6.
For more details about version 4.1.6 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2014/09/kamailio-v4-1-6-released/
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I' trying to register kamailio on SIP Service Provider.
In REGISTER reguest I see that URI is like "sip:DOMAIN:PORT;transport=X",
but my SSP wait in another schema "sip:DOMAIN".
How can I solve it?
Thanks.
Regards,
Alexander Chaliy
mobile: +38 097 102 45 83
mail: a.chaliy84(a)gmail.com
skype: achaliy