Thanks for the input. I'll try the Session Timers and the ka_timer param from the dialog module.
-----Original Message----- From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Olle E. Johansson Sent: Wednesday, October 21, 2015 9:30 AM To: Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org> Subject: Re: [SR-Users] Sending ReINVITE from Kamailio > On 21 Oct 2015, at 09:27, ycaner <yasin.ca...@netgsm.com.tr> wrote: > > Hello; > I think Dialog module can do it with ka_timer. take a look please. > in addition , if you want to know call is still up , check the RTP session. > if there isn't Rtp session , so call is hung up. Asterisk can listen > rtp packet and then in silence it can close session. > > have a look "rtptimeout" parameter > This doesn’t always apply either - if the call is on hold there’s no RTP but it should not be hung up. Asterisk handles this, but for other proxys it’s hard to know the state of the media session. /O _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
