Hi All,
What we want to do is to by-pass the restrictions imposed by different ISPs
etc. which normally block based on the port 5060
The SIP server(including SBC) is a commercial one running on port 5060,
that we cannot change. It is already taking care of the NAT issues.
We want to use Kamailio as the pass-thru proxy in front of the SBC just to
receive the request on a random port, and then forward the request to the
SBC internally being on the same private network.
What would be the best way to do it?
Any help will be highly appreciated.
Rizwan Khan
Hello,
I'm runnig kamailio 4.2.0 and using the lookup_branches function from
the register module to lookup all branches after a alias_db_lookup. The
function seems to work fine but i see the following error in the log
file just after the lookup:
INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment
(0x7fad277fe110/0x7fad277fe148) - ignore
INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment
(0x7fad277fcd38/0x7fad277fcd70) - ignore
INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment
(0x7fad277fcd88/0x7fad277fcdc0) - ignore
Code:
alias_db_lookup("dbaliases");
if (!lookup_branches("location")) {
This only happens when i have multiple branches, could this be a error
in my script?
Thanks,
Jan
Hi All,
I am looking to emulate a local proxy with Kamailio or OpenSips using a UAC
and a UAS (each on separate machines ) but when initiate traffic from UAC
to UAS using SIPp and defining a new socket for each call it does not force
this between proxy and UAS but does between UAC and proxy - I think this
may be a routing configuration?
Thanks
Andrew
Hi Guys,
I'm new on Kamailio and this list, so be patient with me :)
I've built an almost "complete" working SIP server on Ubuntu 14.04 LTS.
I told almost complete because my problem is with carrierroute module because I'm not understanding the routing file and where I should put the code to use function cr_route
I would like to allow SIP calls/video between users for free and send calls to different external Gateways / Switches, etc based on the dialled destination.
In particular, I can send outbound calls a prepaid platform that will allow to bill calls based on CLI validation, but in all my tests, it seems that calls are going out always to the IP address of the remote gateway found into the carrierroute table ignoring the SIP user I'm calling.....what is the correct routing?
I'm finding some docs around there, but some refer to very old versions of Kamailio and some others are not clear to me.
What's wrong with my routing logic?
Thank you so much!
Max
route {
# per request initial checks
route(REQINIT);
# NAT detection
route(NAT);
# handle requests within SIP dialogs
route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing
if (is_method("CANCEL"))
{
if (t_check_trans())
t_relay();
exit;
}
t_check_trans();
# authentication
route(AUTH);
# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
setflag(FLT_ACC); # do accounting
}
# dispatch requests to foreign domains
route(SIPOUT);
### requests for my local domains
# handle presence related requests
route(PRESENCE);
# handle registrations
route(REGISTRAR);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
# dispatch destinations to PSTN
#route(PSTN);
# CARRIERROUTE MODULE routing logic
# check table for carrier default and domain default
if(!cr_route("default", "default", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not allowed");
} else {
# In case of failure, re-route the request
t_on_failure("1");
# Relay the request to the gateway
t_relay();
}
# user location service
route(LOCATION);
route(RELAY);
}
Please,
help me.
I tried several times and i did not find any way to configure kamailio to
start it in stateful mode. Anyone know how to do it?
Thanks,
Luciana.
Hello,
We use get_redirects(*) and t_load_contacts in script.
modparam("tm", "contacts_avp", "$avp(contacts)")
In versions previous to 4.1
print all contacts: $(avp(contacts[*])
delete all contacts: $(avp(contacts[*])=$null
>From 4.1 this avp is changed to xavp structure.
What is the structure name and value name of xavp for print in log
$xavp(contacts=>?) ?
Is it possible to delete xavp?
Thank you,
Julia.
Hello,
I want to announce that a new person got developer GIT write access to
repository: Eloy Coto Pereiro. He has developed a new module named
statsd, which offers a connector from kamailio.cfg to manage values in
statsd. The module will be merge to main repository, now is available at:
- https://github.com/eloycoto/statsd
His git commit id is: eloycoto
My warm welcome and looking forward to future work within the project!
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
(cross posting, looking to get feedback for those with more experience
on github options - playing the lazy guy, not searching much on the web
so far).
In order to move the main git repository to github (where the commits
will be pushed by devs) and have a real-time mirror on our server
(stored very likely on kamailio.org, but we will keep git.sip-router.org
as domain for it), two things have to be sorted out:
1) real time propagation of commits from github to git.sip-router.org
I discovered that github provides a web hook for this event - when a
commit is pushed, then a http query is done to a given url. I thought of
using it to trigger the mirroring from git.sip-router.org
I couldn't find if we can use what Victor Seva configured on
git.sip-router.org to push in real-time to github, which I guess is a
script invoked on commit -- if we can turn that around, would be
probably the easiest.
2) email notifications
Anyone with extensive knowledge of how much the notification messages
can be customized? Eventually trying to get the structure as much as
possible to what we have now -- see an example:
- http://lists.sip-router.org/pipermail/sr-dev/2014-November/026253.html
I could notice that Jitsi (using github) commit notification has a
format not far from ours:
- http://lists.jitsi.org/pipermail/commits/2014-November/012937.html
I can eventually talk with them if no one else here has related knowledge.
Another option is to trigger the notifications from the mirror
repository we will keep on git.sip-router.org
Also, would be good to know if there are people with some spare time
willing to help whenever needed with this migration. Reply here to have
you in mind.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hi list
I don't understand prepaid logic calculation, because when my credit is 0
or lees than cost per second, the call is established equal
$var(i_pulse) and $var(f_pulse) how this works????
