Hi folks,
I discussed this issue in previous email ( [SR-Users Modifying $rU
after lookup() ) but the question is a bit different now.
I would like to change uri like lookup() function does it, but for the
specific user. Problem is I want to have one registration (X) for
several numbers. When this number is called I have to perform lookup()
for this one common registration (X) and send this call there.
To do this I changed $rU just before lookup() for X, and after
lookup() I changed it back for the original number (both changes were
done using variables $rU). However when there are two registrations
(X1, X2), only X1 can be changed for the original number and the
second one (X2) is sent to user without modification.
I also found function reg_fetch_contacts(domain, uri, profile) but in
documentation there is $ulc variable mentioned. This variable is not
documented.
Is there any way how to perform lookup for specific user?
Thanks for the answer.
Regards,
Efelin
Hello
We are using the siptrace module to save the sip-messages to our database.
All the messages are logged except one message. After INVITE, 100 trying
and 200 OK we loose the outbound ACK which is forwarded by Kamailio
statelessly in t_funcs.c. Is there any way to get a siptrace of this
message, too?
I've made a picture of the sip-messages and the lost ACK.
Kind regards
Timo Klecker
I have an error message in kamailio's log repeated many times. I tried to google
it but no results (I think it's forst time I see no results in a google search)
/usr/sbin/kamailio[3769]: ERROR: ctl [binrpc_run.c:942]: ERROR: binrpc:
rpc_printf: binrpc_addstr failed: buffer too small (overflow) (-2)
What does it mean? Any ideas?
thanks,
Jon
Hello Everyone,
I can only imagine how many times this question has come up since post
2008. Please forgive
my reoccurring of the issue.
We are looking to provide carrier grade sip services to our clients
world wide. What we need is a
lightweight, robust and scalable solution that will allow us to
terminate sip calls to our different carriers.
Performance, and high throughput are factors very important to my
employer. Features such as caller
authentication, database back-end, load balancing, and
interoperability with asterisk are things we are
interested in, as was offered using OpenSER.
With three+ open source proxy servers available on the net puts us in
a situation where we have more
solutions to choose from, at the same time wish the features from one
were available in the other, and
vice versa.
With this in mind, we will have to fall back to other factors such as
the most reliable, proven and active
projects. As mentioned, we would choose functional stability over
endless features that we will never use
and that add to the projects fingerprint...
I understand that all three projects are forks from OpenSER, people
would naturally like to know what
differentiates one from the other.
Thanks in Advance,
Nick Khamis
Toronto Hydro Telecom
Hello,
I uploaded my slides presented at Cluecon 2011, last week in Chicago. In
the first part the focus was on what has been happening in the last year
around the project, on the second part showing some example of using it
to secure VoIP networks. The PDF is available at:
* http://asipto.com/u/42
Alexandr Dubovikov presented the recently announced Homer project which
turns Kamailio in a scalable capturing system for SIP traffic, the
slides for this talk are available at:
* http://www.kamailio.org/events/2011-Cluecon/AD-Homerv2.pdf
Stefan Wintermeyer presented Gemeinschaft 4 project, which is targeting
to offer a very secure out-of-the-box IP PBX system, using Kamailio as
one of the major components. The slides will be available in the near
future (I will update the lists) -- he will present the same topic at 10
years SER conference in Berlin:
* http://sip-router.org/10-years-ser/
Cheers,
Daniel
Hello,
based on the last devel IRC meeting, the next major release should
happen sometime by end of September, therefore we should freeze the
development and go into testing phase about one month before.
I propose Monday, August 22, the day to freeze development for version
3.2.0. By then, I hope every developer will be able to push to master
branch what he/she wants to have in 3.2.0. Write here to mailing lists
if you thing about alternatives.
The very good side of 3.2.0 is that we didn't need to touch much the
core and main modules, meaning we have a solid tested foundation,
further testing should be focused on the new modules and the other
enhancements.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi,
my kamailio is sometimes receiving ruri from remote clients which I
assume are not right formatted.
For ex.: sip:user@kamailio=3Buser=3Dphone
Is there any mechanism which could transform this into something
readable for kamailio or do I have to simply cut off everything after
the host name of the RURI ?
Mino
Hi all,
we are trying to setup a kamailio presence server in SIP/SIMPLE domain
to interwork with XMPP domains including the GTalk (see the config
below). The architecture is like this:
- SIP/SIMPLE server: kamailio3.14 with full configuration with XCAP integration
- XMPP Gateway: another kamailio server + ejabberd server
- component mode
- xmpp moduleL: outbound proxy to the Kamailio SIP/SIMPLE server
- pua_xmpp: no outbound proxy?
- XMPP server: ejabberd
The users from XMPP domain can add the SIP account and see the
presence of the SIP users, but not vice versa. The chat from both
directions works fine.
in the Presence Server, we have configured the kamailio with XCAP
integration. We have the following doubts:
- does pua_xmpp/pua modules need the xcap integration for the presence
integration with xmpp domains?
because to use xcap for presence authorization rules, it needs the
sip clients support the xcap
- does pua_xmpp/pua support xcap? otherwise how to works?
- in case of multiple SIP/SIMPLE presence server, how we can configure
the server_address of
pua_xmpp and presence parameter in xmpp gw?
Can you help us to clarify the doubts please?
Many thanks in advanced!
Best Regards,
Laura
PS: following are the main configuration of the xmpp GW:
---------------------------------------------------------------------------------------------
...
modparam("xmpp", "domain_separator", "*")
modparam("xmpp", "backend", "component")
modparam("xmpp", "gateway_domain", "<mygwdomain>")
modparam("xmpp", "xmpp_domain", "<mygwdomain>")
modparam("xmpp", "xmpp_host", "127.0.0.1")
modparam("xmpp", "xmpp_password", "secret")
modparam("xmpp", "outbound_proxy", "<my oubound proxy uri>")
modparam("pua", "outbound_proxy", "<my outbound proxy uri>")
modparam("pua", "update_period", 60)
modparam("pua", "default_expires", 1200)
modparam("presence", "server_address", "<my presence server uri>")
modparam("pua_xmpp", "server_address", "<my presence server uri>")
route{
route(REQINIT);
t_check_trans();
if (uri =~ "sip:.+@.*<myxmppdomain>") {
route(PRESENCE);
if ($rU==$null)
{
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
route(CHAT);
}
xlog("L_INFO", "*** xmpp: unhandled message type\n");
t_reply("503", "Service unavailable");
return;
}
...
route[CHAT] {
if(!is_method("MESSAGE"))
return;
if (!t_newtran()) {
sl_reply_error();
exit;
}
xlog("L_INFO", "*** xmpp-handled MESSAGE message.");
if ($cT=~"^text/plain") {
if (xmpp_send_message()) {
t_reply("200", "Accepted");
} else {
t_reply("404", "Not found");
}
} else {
xlog("L_INFO", "*** xmpp-handled MESSAGE, ignoring not text messages");
t_reply("200", "Accepted");
}
t_release();
exit;
}
route[PRESENCE] {
if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY"))
return;
# create new transaction to catch retransmissions
if (!t_newtran())
{
sl_reply_error();
exit;
}
if( is_method("NOTIFY"))
{
xlog("L_INFO", "*** xmpp-handled NOTIFY message.");
if(pua_xmpp_notify())
t_reply("200", "OK");
t_release();
exit;
} else if( is_method("SUBSCRIBE"))
{
xlog("L_INFO", "*** xmpp-handled SUBSCRIBE message.\n");
handle_subscribe();
if($(hdr(Event))== "presence")
{
pua_xmpp_req_winfo("$ruri", "$hdr(Expires)");
}
t_release();
exit;
} else if(is_method("PUBLISH"))
{
handle_publish();
t_release();
exit;
}
}
...
Hello
When one creates a user using the cli command like this: "kamctl add 200 abc", it is my understanding that the information goes to the subscriber table and into memory. Is that right?
But if one populates the subscriber table manually [adding a record on the table w/o the /kamctlcli command], do we need to "load" those accounts into memory before a client tries to register or does kamailio looks it up when the registration request comes in at that time?
What we are trying to do is to bulk load subscribers on the subscribers table instead of using the "kamctl add" command. For example, with LCR parameters we modify the tables manually and then do a "lcr_reload" and all the changes will be in memory after the reload.
thank you in advance
fborot
