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From: *Daniel-Constantin Mierla* <mico...@gmail.com
<mailto:mico...@gmail.com>>
Date: Wed, Aug 12, 2009 at 9:20 AM
Subject: Re: [Kamailio-Users] dialog info + Grandstreams = freezing
To: David <kamailio.org <http://kamailio.org>@spam.lublink.net
<http://spam.lublink.net>>
Hello,
On Tue, Aug 11, 2009 at 2:18 PM, David <kamailio.org
<http://kamailio.org>@spam.lublink.net <http://spam.lublink.net>> wrote:
Hey,
I am not good at diagrams, but here goes :
[ Asterisk] [ Asterisk ] ================> [ MySQL ]
/ \ / \
[ Kamailio ] ============/\
/ \ / \ / \
[ Extension ] [ Extension ] [ Extension ]
Here is a simple flow :
1. Extension 1 is plugged in
2. It registers to Kamailio and Kamailio stores to Location
3. Extension 1 subscribes to all the other extensions
4. Kamailio calls handle_publish() and t_release()
5. Extension 1 dials '102'
6. Kamailio receives the INVITE chooses an asterisk server and
relays the INVITE ( t_relay )
7. Asterisk does some checking, starts monitor ( if needed ) and
some other stuff
8. Asterisk sends INVITE to Kamailio for testspace.102
9. Kamailio finds testspace.102 in database and forwards INVITE to
the appropriate extension
10. Extension replies 'OK'
11. Kamailio, using reply-route, sends it back to asterisk
12. Asterisk bridges the call with 101
13. Asterisk sends an OK back through Kamailio to Extension 1.
14. Asterisk tries to reinvite the audio ( if not monitoring )
15. Audio bypasses Asterisk, but the SIP still is making the full
trip.
Does this answer your question?
Do you do call tracking with dialog for both legs (inbond: phone to
asterisk; outbound: asterisk to phone)? You should do it for one,
outbound is the best, in case asterisk auto-answers the inbound leg.
Cheers,
Daniel
David
Daniel-Constantin Mierla wrote:
Hello,
I am using the module with snom 370 and works ok. The BLF is
on for Polycoms but they do not implement the call pickup
properly (they have direct call pickup as far as I could get).
Please take latest kamailio 1.5 from svn branch 1.5 as there
were committed some updates (maybe they don't affect you,
being mainly to rls, but is better to have the latest stable
so we refer to same source code).
What would be good is to have a diagram of the signaling,
since you say there are lot of messages sent around.
http://callflow.sourceforge.net/
Cheers,
Daniel
On 11.08.2009 5:03 Uhr, David wrote:
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....@myserver to 102 ( this is sent to
asterisk )
sip:testspace....@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 :
r...@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
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