[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

[Jonathan H]

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 <andregronwald78 at 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:2636146e0 at sipgate.de:5060', retrying in '60'
>     -- Contact pjsip_sipgate/sip:2636146e0 at 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