11 Aug
2009
11 Aug
'09
6:03 a.m.
Hello,
I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04,
1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS
1.5.2-tls compilde on Ubuntu 8.04
I am trying to setup presence_dialoginfo with my Grandstreams, Snom and
Linksys. I have a 4 phones on the server.
101 - Linksys SPA962 ( 6.1.5a )
102 - Grandstream GXP2020 ( 1.2.1.4 )
103 - Grandstream GXP2000 ( 1.2.1.4 )
104 - Grandstream GXP2000 ( 1.1.6.46 )
105 - Snom 360 ( 7.3.23 )
My Kamailio deals with registrations, NAT and BLF everything else is
sent to one of two asterisk boxes. I use the dispatcher module for this.
This means that when I call one extension to the other, both call legs
from asterisk are going through Kamailio as separate calls. But to
divide my customers, the usernames are different from the URL that the
user types. For example the customer dials '101' but it is changed to
testspace.101 when it comes back from asterisk. So Kamailio would have
two calls in the event that 101 dials 102.
sip:testspace.101@myserver to 102 ( this is sent to asterisk )
sip:testspace.101@myserver to testspace.102 ( this is coming back from
asterisk )
Something is horribly wrong. I have the following problems :
1. If 102 calls 103, when 103 answers both phones hang for about 2 minutes
2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on
the Linksys and dark on the Snom), but the orange light stays on on the
Snom and it thinks the call is still active ( the light is on, but the
call is over )
3. If any extension calls any extension and I try a call pickup, it
fails. It looks like the Linksys is sending a NOTIFY to pickup the call
( I thought it was supposed to send an invite... ? )
Looking at the logs it looks like Kamailio is sending out so many
NOTIFYs that it is crashing the Grandstreams, and causing the Snom to
act funny.
Here are some experts from my config file :
root@kamailio-dev:/etc/kamailio# grep dialog *
kamailio.cfg:# * avp value for dialogs is still not correct
kamailio.cfg:loadmodule "dialog.so"
kamailio.cfg:loadmodule "presence_dialoginfo.so"
kamailio.cfg:loadmodule "pua_dialoginfo.so"
kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0)
kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0)
kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0)
kamailio.cfg:modparam("dialog", "dlg_flag", 4)
kamailio.cfg:modparam("dialog", "db_mode", 1)
kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)")
# I still
haven't figured out how to set $avp(i:10)
kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300)
kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog",
1)
kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1)
kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog",
"db_url",
kamailio.cfg:# Flag 4 = Mark the current request for a dialog
kamailio.cfg: # sequential request withing a dialog should
the set flag looks like this :
if ( ds_is_from_list() )
{
xlog("L_INFO", "Coming from asterisk");
if ( is_method("INVITE"))
{
setflag(4);
}
}
So the dialog flag is only set for the leg coming back from asterisk.
When a notify comes in :
if(is_method("NOTIFY") )
{
if (! t_newtran())
{
sl_reply_error();
exit;
};
t_reply("200", "OK");
t_release();
exit ;
}
Publish and subscribe are like this :
if( is_method("PUBLISH") || is_method("SUBSCRIBE") )
{
route(5);
exit;
}
route[5]
{
# absorb retransmissions
if (! t_newtran())
{
xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition -
M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_reply_error();
exit;
};
append_to_reply("Contact: <sip:myserver.tld:5060>\r\n");
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
} else if( is_method("SUBSCRIBE")) {
handle_subscribe();
t_release();
}
else
{
}
exit;
}
I also have NAT checking for those telephones where stun isn't enough.
Before I reach publish/subscribe/invite/notify, I also call setbflag()
and sometimes call fix_nated_contact(). Additionnally, I have a block if
code before my presence stuff if ( has_totag() && loose_route()) {
t_relay(); }.
If sip.conf:canreinvite=yes, the grandstreams freeze so long that the
server times out, and the BLFs get really messed up. if
sip:canreinvite=no the grandstreams only freeze for about 30 seconds.
Obviously I am doing something wrong, but despite having searched google
for endless hours, and poured over documentation, I can not seem to find
what I did wrong.
I would really appreciate if someone could shed light on my problem.
I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04,
1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS
1.5.2-tls compilde on Ubuntu 8.04
Thanks,
David