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

[Jonathan H]

Hmmm, sorry, I can't think of anything except... why do you need the
STUN server? And are you sure that all the ports in your router
definitely match the ones Asterisk thinks it's using?

Then there is always the SIP-ALG problem with some routers, which some
people have been able to overcome by switching to TLS, and I see that
SIPgate offer TLS.
You could try making a free certificate and going TLS which uses port
5061. No promises, but worth a try as it fixed the issue for a
different poster.

The only other thing I can find while Googling for this, which solved
it for someone else, was related to DNS server issues, but this seems
unlikely (although not impossible).

On 15 October 2016 at 10:07, Andre Gronwald <andregronwald78 at gmail.com> wrote:
> ping times are fine as well:
>
> [root at freepbx asterisk]# ping sipgate.de
> PING sipgate.de (217.10.79.9) 56(84) bytes of data.
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57 time=46.8 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=5 ttl=57 time=47.1 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=6 ttl=57 time=46.4 ms
> 64 bytes from sipgate.de (217.10.79.9): icmp_seq=7 ttl=57 time=47.1 ms
> ^C
> --- sipgate.de ping statistics ---
> 7 packets transmitted, 7 received, 0% packet loss, time 6360ms
> rtt min/avg/max/mdev = 46.406/46.809/47.191/0.393 ms
> [root at freepbx asterisk]#
>
>
> this high RTT appears only sometimes.  After removing STUN-server it looks
> better, did two test calls right now, both gone through immediately. At the
> end of the second test call I see:
>
>     -- Executing [s at app-announcement-1:5]
> Playback("PJSIP/pjsip_sipgate-00000003",
> "custom/araz01&custom/07-polly,noanswer") in new stack
>     -- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/araz01.alaw' (language
> 'en')
>     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Reachable.
> RTT: 493.094 msec
>   == Endpoint pjsip_sipgate is now Reachable
>     -- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/07-polly.slin'
> (language 'en')
>     -- Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now
> Unreachable.  RTT: 0.000 msec
>   == Endpoint pjsip_sipgate is now Unreachable
>
>
> Why do I have that loss of registrations?
>
> here my pjsip config for sipgate.de:
>
> freepbx*CLI> pjsip show registration pjsip_sipgate
>
>  <Registration/ServerURI..............................>  <Auth..........>
> <Status.......>
> ==========================================================================================
>
>  pjsip_sipgate/sip:sipgate.de:5060                       pjsip_sipgate
> Registered
>
>  ParameterName            : ParameterValue
>  ========================================================
>  auth_rejection_permanent : true
>  client_uri               : sip:2636146e0 at sipgate.de:5060
>  contact_user             : 2636146e0
>  endpoint                 :
>  expiration               : 600
>  fatal_retry_interval     : 0
>  forbidden_retry_interval : 0
>  line                     : false
>  max_retries              : 10
>  outbound_auth            : pjsip_sipgate
>  outbound_proxy           :
>  retry_interval           : 60
>  server_uri               : sip:sipgate.de:5060
>  support_path             : false
>  transport                : 0.0.0.0-udp
>
> Remind: Endpoint is currently unreachable, but asterisk shows "Registered".
> Test call fails at this moment.
>
>
> regards,
> andre
>
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