[OpenSIPS-Users] [Kamailio-Users] dialog info + Grandstreams = freezing
David
kamailio.org at spam.lublink.net
Tue Aug 11 14:18:01 CEST 2009
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?
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.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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