thr3ads.net - asterisk users - [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore [Oct 2016]

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Andre Gronwald

2016-Oct-15 09:07 UTC

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

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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Jonathan H

2016-Oct-15 09:25 UTC

head link

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

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
>
> --
> _____________________________________________________________________
> -- 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:
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Andre Gronwald

2016-Oct-15 10:18 UTC

head link

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

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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Andre Gronwald

2016-Oct-15 11:05 UTC

head link

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

hi,
let me explain in detail, what i have configured and what is happening now:

1st router w724v (Deutsche Telekom AG):
     - port forwarding, everything to destination port 51000-55999 to 
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
     - configured NAT, everything to port 51000-55999 to device 
192.168.3.99 (same ports)

other direction is totally open.

I observed that all sip calls are closed exactly after 32s. call is 
disconnected on calling side as well... seems to be a timeout issue.

here i have some debug logs. I see lot of requests from asterisk to 
sipgate.de, which are not answered. but communication is going fine in 
both directions (otherwise registration would not be possible?):


<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0 at 80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "02363361779" <sip:02363361779 at
sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9 at 217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0 at 80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9 at 217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

     -- Executing [s at app-announcement-1:3] 
Wait("PJSIP/pjsip_sipgate-00000003", "1") in new stack
        > 0x7f2ee8037810 -- Probation passed - setting RTP source 
address to 192.168.2.1:7070
     -- Executing [s at app-announcement-1:4] 
NoOp("PJSIP/pjsip_sipgate-00000003", "Playing announcement ARAZ 
(Au?erhalb Regelarbeitszeit)") in new stack
     -- 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')
<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

     -- <PJSIP/pjsip_sipgate-00000003> Playing
'custom/07-polly.slin'
(language 'en')
<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (429 bytes) to UDP:217.10.79.9:5060 --->
OPTIONS sip:2636146e0 at sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:2636146e0 at sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:2636146e0 at sipgate.de>
Contact: <sip:2636146e0 at 80.142.13.32:55060>
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Transmitting SIP request (429 bytes) to UDP:217.10.79.9:5060 --->
OPTIONS sip:2636146e0 at sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:2636146e0 at sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:2636146e0 at sipgate.de>
Contact: <sip:2636146e0 at 80.142.13.32:55060>
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP response (338 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:2636146e0 at sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:2636146e0 at
sipgate.de>;tag=065a2aa3915c789dd1a0ab4f12b0002c.4434
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Content-Length: 0


<--- Received SIP response (338 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:2636146e0 at sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:2636146e0 at
sipgate.de>;tag=065a2aa3915c789dd1a0ab4f12b0002c.4434
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Content-Length: 0


<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0
Via: SIP/2.0/UDP 
172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0
Via: SIP/2.0/UDP 
217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
To: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
<--- Transmitting SIP request (543 bytes) to UDP:217.10.79.9:5060 --->
BYE sip:0xxxxxxxx9 at 217.10.77.115:5060 SIP/2.0
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjmhpnZAJdsqBV9w-4WA.1DjZHqFpj6-au
From: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
To: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
CSeq: 5732 BYE
Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Route: <sip:172.20.40.6;lr>
Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


freepbx*CLI>
<--- Received SIP response (446 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjmhpnZAJdsqBV9w-4WA.1DjZHqFpj6-au
From: <sip:2636146e0 at sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
To: "0xxxxxxxx9" <sip:0xxxxxxxx9 at sipgate.de>;tag=as02fa8fcc
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49 at sipgate.de
CSeq: 5732 BYE
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (545 bytes) to UDP:217.10.79.9:5060 --->
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjShmmpRhUENHI8CUFtNiZttd1lZohqw6p
From: <sip:2636146e0 at sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:2636146e0 at sipgate.de>
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55530 REGISTER
Contact: <sip:2636146e0 at 80.142.13.32:55060>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP response (436 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjShmmpRhUENHI8CUFtNiZttd1lZohqw6p
From: <sip:2636146e0 at sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:2636146e0 at
sipgate.de>;tag=86e53dd608d1c001e0b8060625977563.c38e
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55530 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="WAIJSVgCCB1kfjXwrwmT7mfxLr/nkdQO"
Content-Length: 0


<--- Transmitting SIP request (723 bytes) to UDP:217.10.79.9:5060 --->
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjkC0dwtjcOsKzwskJq2gE2RelAFlFm7cw
From: <sip:2636146e0 at sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:2636146e0 at sipgate.de>
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55531 REGISTER
Contact: <sip:2636146e0 at 80.142.13.32:55060>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, 
UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Authorization: Digest username="2636146e0",
realm="sipgate.de",
nonce="WAIJSVgCCB1kfjXwrwmT7mfxLr/nkdQO",
uri="sip:sipgate.de:5060",
response="514fd5c1b4aa1b951400836d2b5a0b10"
Content-Length:  0


<--- Received SIP response (395 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
80.142.13.32:55060;rport;branch=z9hG4bKPjkC0dwtjcOsKzwskJq2gE2RelAFlFm7cw
From: <sip:2636146e0 at sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:2636146e0 at
sipgate.de>;tag=86e53dd608d1c001e0b8060625977563.2957
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55531 REGISTER
Contact: <sip:2636146e0 at 80.142.13.32:55060>;expires=60
Content-Length: 0







kind regards,
andre

asterisk users - Oct 2016 - Registered successfully, but after a minute or so no SIP messages anymore

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

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

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

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