[Kamailio-Users] dialog info + Grandstreams = freezing

[David]

I installed kamailio 1.5 from branch without any difference.

But now we refer to the same code.

David

> Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> 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).
>>
>> 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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>>>
>>
>
>
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