Hi,
I am trying to use GRUU for routing MESSAGEs. For example
if I want to route to specific instance of the registered user, what do I
need
to send in the MESSAGE? If I know pub-gruu of the instance, where do I
insert it so that kamailio can correctly process it and send it to the
right instance?
Thanks
Krish Kura
Hi,
I have a kamailio installation running, but in kamctl the moni command
works, but stop, restart, start command doesn't work...
why is it so??
Regards,
Vineet Menon
Hi!
For example:
$avp(i:test) = 1;
$avp(i:test) = 2;
# now: test[0]=2, test[1]=1
I would think that assigning or deleting using an index would overwrite
the respective element in the AVP list, but assigning $null always
deletes [0] and assigning a value always add another AVP to the top, e.g:
$(avp(i:test)[1]) = 3;
# I would expect: test[0]=2, test[1]=3
# but it is: test[0]=3, test[1]=2, test[2]=1
$(avp(i:test)[2]) = $null;
# I would expect: test[0]=3, test[1]=2
# but it is: test[0]=2, test[1]=1
So, is the current behavior really the correct behavior? It is not what
I would expect when using indexes.
If it is correct, I will improve the documentation accordingly.
regards
Klaus
Hello,
several days ago, just before freezing the development for v3.3.0, I
added to dialog module the option to send keep alives for ongoing calls
in order to detect if caller/callee is gone.
SIP specs require to increment the CSeq for requests within dialog, but
because the keep alives are sent from the sip server, it would results
de-synchronization of the CSeq values hold in phones themselves (e.g., a
BYE created by caller/callee after a keep alive will be with lower cseq
than the other side would expect and accept). One solution would be to
update cseq when BYE passes the server to the right value. This implies
more processing, as a call can have many re-INVITEs or other requests
within dialog sent by caller/callee, including periodic updates to
database to store cseq numbers.
So I went for a different approach, like stated in subject -- let's see
your opinion if you think is going to work. Practically the keep alives
will be OPTIONS with CSeq equal or lower than the last valid value
(e.g., the cseq of the INVITE creating the call).
If the caller/callee is gone, that is simply, the OPTION will be timed
out and dialog module will send BYE.
If the caller/callee are reachable, but for some reason the call was
destroyed (e.g., a crash and restart meanwhile), since there is a
To-tag, the OPTIONS should get a 481 Call/Leg transaction does not
exists. Again, a case when kamailio will end the dialog.
The one to be discussed here would be caller/callee are still on the call.
Based on RFC (and the feedback from people on another thread here), a
requests coming within a dialog with lower CSEq should be replies with
500. CSeq numbers are still ok in both sides (note that requests with
lower CSeq can happen in reality, like two fast re-INVITEs sent over
UDP, the second arriving first due to network transmission).
I tested with snom phones and jitsi so far, seems to be a working
solution (well, jitsi was replying 500 after 20secs to keep alive
OPTIONS request, not sure for what reason, just reported back to the
project).
Is anyone here seeing any possible issues with the approach?
Call are ended by dialog, with proper CSeq in the BYE, after 10seconds
from the moment 408 or 481 is received, no other replies are taken in
consideration for ending the dialog from server side.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I have problem with loading perl module on Kamailio v3.2 installed from
repository on Debian sqeeze amd64. Perl modules was also installed from
repository package kamailio-perl-modules. I have also installed all perl
dependences listed in perl module documentary.
Into Kamailio config file I added:
loadmodule "perl.so"
and
modparam("perl", "filename", "/etc/kamailio/skript.pl")
Now when I'm trying to start Kamailio I'll get following:
May 1 23:13:46 server kamailio: DEBUG: <core> [cfg.y:1712]: loading module
perl.so
May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:557]:
load_module: trying to load </usr/lib64/kamailio/modules_k/perl.so>
May 1 23:13:46 server kamailio: WARNING: <core> [sr_module.c:620]:
/usr/lib64/kamailio/modules_k/perl.so: exports dlflags interface is
deprecated and it will not be supported in newer versions; consider using
mod_register() instead
May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:96]:
set_mod_param_regex: 'perl' matches module 'perl'
May 1 23:13:46 server kamailio: DEBUG: <core> [sr_module.c:763]:
find_param_export: found <filename> in module perl
[/usr/lib64/kamailio/modules_k/perl.so]
May 1 23:13:46 server kamailio: DEBUG: <core> [modparam.c:113]:
set_mod_param_regex: found <filename> in module perl
[/usr/lib64/kamailio/modules_k/perl.so]
May 1 23:13:46 server /usr/sbin/kamailio[31192]: DEBUG: <core>
[sr_module.c:928]: DEBUG: init_mod: perl
May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: perl [perl.c:238]:
failed to load perl file "/etc/kamailio/skript.pl".
May 1 23:13:46 server /usr/sbin/kamailio[31192]: ERROR: <core>
[sr_module.c:932]: init_mod(): Error while initializing module perl
(/usr/lib64/kamailio/modules_k/perl.so)
I am confused about the WARNIG too, can it have some influence to the
errors?
File skript.pl is in the specified directory. I also changed permissions to
777. It's content is from sample "branches.pl".
I have similar problem with module app_python.so.
Can you please give me some advice? It's important for me to get this
module working.
Thank you.
--
Regards Ladislav Jurak
Hi, I'm using Siremis 3.2 ans Kamailio 3.2.3 for my final project of my
bachloor degree
I would like to know how Siremis and Kamailio work together? Like:
If I decide to set a protocol in the acl trusted in Siremis like TLS, ok,
it works well !!
But I need to know the different links between siremis and kamailio to
working well.. Why?
Cause I need to implement in Siremis a choise form about the sdpops module,
and it has to work well of course with the database of ser!
So.. What are the different steps to do this?
- Create new table sdpops in ser
- create the form in siremis and view ..
- what in kamailio.cfg??
I found this: http://kb.asipto.com/siremis:install32x:new-views
OK, but in my case, I need to do that for SDPOPS, which is a module of
kamailio, whithout any table and components about!
I mean, there is no openser.sdpops in the database of kamailio, so create
the table ok.. but what are the columns, datas, to set in ??
cause I tried php gen_meta.php Serdb sdpops ser.acl.sdpops of course it
does not work --> no openser.sdpops ......
thx in advance for your help
Best regards
--
*Grégoire Vandendeurpel, *
*
*
*IT Sector*
Greetings,
I am confused at some functionality I am seeing with Kamailio 1.5.4. I
know this is an old version, but I don't have the time to go through a
lengthy upgrade process right now. The issue I am seeing is that the
server is inserting a Route header with it's own IP address for an unknown
reason. Here is the initial invite (removed SDP for simplicity):
INVITE sip:13@67.207.130.146:5060 SIP/2.0
Via: SIP/2.0/UDP 68.64.220.108:5060;branch=z9hG4bK78dd33c6;rport
From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
>;tag=as1cad6370
To: <sip:13@67.207.130.146:5060>
Contact: <sip:9546496707@68.64.220.108>
Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 70
Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Route: <sip:13@boulder-voip.mydomain.com>
P-Account-ID: 99990023
P-Proxy-Route: Yes
Content-Type: application/sdp
Content-Length: 240
The basics of what happen next are:
t_check_trans();
record_route();
remove_hf("P-Proxy-Route");
if(loose_route()){
route(3);
}
route[3]{
t_on_reply("1");
if(!t_relay()){
sl_reply_error();
}
}
The INVITE that goes out has the funky Route: header with the Kamailio IP
in there. This is causing problems for some of the upstream proxy servers
(obviously).
INVITE sip:13@boulder-voip.mydomain.com SIP/2.0
Record-Route: <sip:67.207.130.146;lr;ftag=as1cad6370>
Via: SIP/2.0/UDP 67.207.130.146;branch=z9hG4bKf183.456d51e1.0
Via: SIP/2.0/UDP 68.64.220.108:5060
;received=68.64.220.108;branch=z9hG4bK78dd33c6;rport=5060
From: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
>;tag=as1cad6370
To: <sip:13@67.207.130.146:5060>
Contact: <sip:9546496707@68.64.220.108>
Call-ID: 43134ece101abfca6ecab20212295909(a)dev-asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: G-Tel v1.0
Max-Forwards: 69
Remote-Party-ID: "WIRELESS CALLER" <sip:9546496707@dev-asterisk.mydomain.com
>;privacy=off;screen=no
Date: Tue, 01 May 2012 18:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
P-Account-ID: 99990023
Content-Type: application/sdp
Content-Length: 240
Route: <sip:13@67.207.130.146:5060>
Any idea what may be causing this to happen and how I could prevent it? I
have tried removing the Route header using the remove_hf("Route") before
doing the t_relay, but that doesn't seem to help.
Thanks,
Geoff
Hi,
I have a ubuntu 12.04 installation with me, and want to install kamailio in
it... I have added repo from kamailio.ord site, but they are pretty old and
require libmysqlclient16, which is no longer available.
Ubunut 12.04 has a canonical repo for openser.... are they both
interchangeble??
can i install openser and feel the same as kamailio??
Regards,
Vineet Menon
Hello,
Does anyone can give me a proper example how setup LCR to be able to use
prefix and from_uri field.
Does not a route should be preferred if prefix or from_uri is more accurate
? I mean that if you st from_uri value instead of default NULL route
should be considered as better ?
Thank you for you help
Dear list,
we are running Kamailio 3.2 together with the included OpenXcap server.
I try to retrieve only part of a certain resource list using XPath, but the
server always returns the full list, see example below. I was so far not
able to retrieve only part of a resource list, independent of the XPath
expression I'm using.
Do I make something wrong or is this a bug?
Thanks and regards,
Fabian
fubeh@orion:~$ curl --digest -u test4:XXXX
http://192.168.1.134:5060/xcap-root/resource-lists/users/sip:test4@192.168.…
<?xml version="1.0"?>
<resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists">
<list name="Business">
<entry uri="sip:test1@192.168.1.134">
<display-name>Test 1</display-name>
</entry>
<entry uri="sip:test122@192.168.1.134">
<display-name>Test 1</display-name>
</entry>
</list>
<list name="Family">
<entry uri="sip:test2@192.168.1.134">
<display-name>Test 2</display-name>
</entry>
<entry uri="sip:test222@192.168.1.134">
<display-name>Test 2</display-name>
</entry>
</list>
<list name="Friends">
<entry uri="sip:test3@192.168.1.134">
<display-name>Test 3</display-name>
</entry>
<entry uri="sip:test322@192.168.1.134">
<display-name>Test 3</display-name>
</entry>
</list>
<list name="Blocked" />
</resource-lists>
