[Kamailio-Users] dialog info + Grandstreams = freezing
Daniel-Constantin Mierla
miconda at gmail.com
Tue Aug 11 09:21:01 CEST 2009
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.101 at myserver to 102 ( this is sent to asterisk )
> sip:testspace.101 at 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 at 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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>
--
Daniel-Constantin Mierla
* SIP Router Bootcamp
* Kamailio (OpenSER) and Asterisk Training
* Berlin, Germany, Sep 1-4, 2009
* http://www.asipto.com/index.php/sip-router-bootcamp/
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