Hello, Yes, I am tracking both legs. Because, if a call comes in on a DID, Asterisk decides which extension to ring and it's the second leg which needs to be tracked, whereas if it is an outbound call to a PRI, it's the first leg that should be tracked because the second leg doesn't make it through Kamailio ( for now ). So you think my problem is because I am tracking both legs? I thought of this last night as well, I'll have a look today or tomorrow and will post my results.

David Daniel-Constantin Mierla wrote: > .. fwd to list ... reply all button mismatch ... > > ---------- Forwarded message ---------- > From: *Daniel-Constantin Mierla* &lt;miconda(a)gmail.com > <mailto:miconda@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.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 > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users