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

[Andre Gronwald]

Thanks Jonathan for your support.

I would like to avoid TLS at the moment  (in general I am a fan of 
secured communication!) because the other provider is not supporting 
TLS. And sipgate is just used for testing.

However I can see the following which is quite interesting:

[2016-10-15 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Reachable. 
RTT: 433.814 msec
[2016-10-15 11:20:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:21:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Reachable. 
RTT: 439.006 msec
[2016-10-15 11:30:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:31:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Reachable. 
RTT: 433.426 msec
[2016-10-15 11:40:30] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Reachable
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Contact pjsip_sipgate/sip:2636146e0 at sipgate.de:5060 is now Unreachable. 
RTT: 0.000 msec
[2016-10-15 11:41:22] VERBOSE[14791] res_pjsip/pjsip_configuration.c: 
Endpoint pjsip_sipgate is now Unreachable

I think that the times are matching exactly the qualify frequency and 
registry expiration - expiration is set to 600s, and qualify frequency 
to 50s. Seems that the qualify requests are not supported (this is the 
case for the other provider as well!). So maybe I should work without 
sip qualify.

Besides this I have another curiousity:
One call:
     -- Executing [s at app-announcement-1:3] 
Wait("PJSIP/pjsip_sipgate-00000019", "1") in new stack
        > 0x7fabf004bfd0 -- Probation passed - setting RTP source 
address to 217.10.77.109:16248
Another call:
     -- Executing [s at app-announcement-1:3] 
Wait("PJSIP/pjsip_sipgate-0000001a", "1") in new stack
        > 0x7fabf0070bb0 -- Probation passed - setting RTP source 
address to 192.168.2.1:7074

??? 217.10.77.109 is sipgate.de -> ok. 192.168.2.1 is my 
vDSL-access-router ??? Why does the RTP source address changes? that 
must not happen.


And another observation: I am registered to sipgate.de, fine. Incoming 
call is processed,  announcement is played. But when the caller hangs up 
asterisk is not recognizing it. it takes about 16s until the channel is 
closed after hangup?


regards,
andre
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