All other things aside, this stands out immediately: RTT: 434.393 msec
That's almost half a second round trip for a packet. I'm amazed anything works at all. For SIP connections, mine are usually about 26ms max, anything above about 35 is bad. Looks like a serious config issue. Try pinging and see what you get - my ping times to sipgate.de from the UK are Best:13.6ms Worst 13.8ms across 100 pings. I could be wrong, but I'd be surprised if that wasn't causing problems, at least with audio. On 15 October 2016 at 09:11, Andre Gronwald <andregronwal...@gmail.com> wrote: > Hi all, > I have an issue with asterisk 13 and pjsip. I guess it is somehow Firewall > related, but I'm unsure. > > A registration to Sipgate is established successfully: > > > <Registration/ServerURI..............................> <Auth..........> > <Status.......> > ========================================================================================== > > pjsip_sipgate/sip:sipgate.de:5060 pjsip_sipgate > Registered > > > Calling the registered number is even successfully shown in asterisk (it is > a freepbx installation). > But when doing a second call the number is busy ("provider" busy, I don't > see anything in asterisk verbose mode). > Sending a pjsip unregister results in the following messages: > > [2016-10-15 10:03:22] WARNING[10162]: res_pjsip_outbound_registration.c:761 > schedule_retry: No response received from 'sip:sipgate.de:5060' on > registration attempt to 'sip:263614...@sipgate.de:5060', retrying in '60' > -- Contact pjsip_sipgate/sip:263614...@sipgate.de:5060 is now Reachable. > RTT: 434.393 msec > == Endpoint pjsip_sipgate is now Reachable > > so it is somewhat clear, why i get a busy, because the endpoint is not > reachable. But WHY is the endpoint not reachable? > > Regarding the architecture: I have two routers cascaded, that is > unfortunately necessary. On the first router (vDSL-access router) I have > forwarded nearly everything to the second router (Bintec rj 353), where a > port forwarding for relevant ports (sip and pjsip (udp and tcp), rtp (udp)) > is configured. IF a call goes through, nearly everything is working (audio > only incoming, but that is another issue). > > STUN is configured. FreePBX Firewall is disabled. > > Kind regards, > andre > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 > http://www.asterisk.org/community/astricon-user-conference > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users