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:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]: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:[email protected]: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@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@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
--
_____________________________________________________________________
-- 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
