Registered Successfully, But After A Minute Or So No SIP Messages Anymore

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Asterisk Users 11 Comments

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:



=========================================================================================
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@sipgate.de:5060′, retrying in ’60’
— Contact pjsip_sipgate/sip:2636146e0@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

11 thoughts on - Registered Successfully, But After A Minute Or So No SIP Messages Anymore

  • Very interesting: I have another provider configured, that was not reachable as well. I disabled the STUN-server (external STUN server), and now the second registration works fine, BUT with the same “error”
    messages (unreachable etc) as the other provider. But in contrast the number is always reachable!!!

    Is there any explanation for this? I just want to understand… 😉 … and solve it.

    regards, andre

    Am 15.10.2016 um 10:11 schrieb Andre Gronwald:

  • 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.

  • ping times are fine as well:

    [root@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 ttlW timeF.7 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttlW timeF.4 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttlW timeF.7 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttlW timeF.8 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=5 ttlW timeG.1 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=6 ttlW timeF.4 ms
    64 bytes from sipgate.de (217.10.79.9): icmp_seq=7 ttlW timeG.1 ms
    ^C
    — sipgate.de ping statistics –

  • 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).

  • 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@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@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@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@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@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@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@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

  • 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@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:
    Record-Route:
    Record-Route:
    From: “02363361779” ;tag=as02fa8fcc To:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de CSeq: 103 INVITE
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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@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:
    Record-Route:
    Record-Route:
    From: “0xxxxxxxx9” ;tag=as02fa8fcc To:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de CSeq: 103 INVITE
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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@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@app-announcement-1:4]
    NoOp(“PJSIP/pjsip_sipgate-00000003”, “Playing announcement ARAZ
    (Außerhalb Regelarbeitszeit)”) in new stack
    — Executing [s@app-announcement-1:5]
    Playback(“PJSIP/pjsip_sipgate-00000003”,
    “custom/araz01&custom/07-polly,noanswer”) in new stack
    — 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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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

    — 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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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@sipgate.de:5060 SIP/2.0
    Via: SIP/2.0/UDP
    80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
    From: ;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
    To:
    Contact:
    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@sipgate.de:5060 SIP/2.0
    Via: SIP/2.0/UDP
    80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
    From: ;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
    To:
    Contact:
    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: ;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
    To: ;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: ;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
    To: ;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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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:
    Record-Route:
    Record-Route:
    Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de From: “0xxxxxxxx9” ;tag=as02fa8fcc To: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf CSeq: 103 INVITE
    Server: FPBX-13.0.188.8(13.11.2)
    Contact:
    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@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: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf To: “0xxxxxxxx9” ;tag=as02fa8fcc Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@sipgate.de CSeq: 5732 BYE
    Route:
    Route:
    Route:
    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: ;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf To: “0xxxxxxxx9” ;tag=as02fa8fcc Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7d49@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: ;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
    To:
    Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu CSeq: 55530 REGISTER
    Contact:
    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: ;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
    To: ;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: ;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
    To:
    Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu CSeq: 55531 REGISTER
    Contact:
    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: ;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
    To: ;tag=86e53dd608d1c001e0b8060625977563.2957
    Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu CSeq: 55531 REGISTER
    Contact: ;expires=60
    Content-Length: 0

    kind regards, andre

    —

  • Hi,

    I don’t see any SIP ACK’s in your trace.

    Is the SIP 200 OK reaching the originating caller, or being blocked on the way through?

    Asterisk will tear down the call after ~30secs of audio playing in both directions if it doesn’t receive the SIP ACK.

    Regards,

    Ian

    —

  • ok, solved the firewall issue. A first test call worked fine. Another one now still gets disconnected after 32s.

    But in FW there are no blocked packets anymore?!

    And I don’t understand why the registration to the same IP and same Port is working, but not later transmission of further SIP packets? that doesn’t sound logical to me. What do you think?

    regards, andre

  • Have you tried setting keepalive(20 seconds) on your sip.conf and on your phones ?

    From: Andre Gronwald
    To: asterisk-users@lists.digium.com Sent: Saturday, October 15, 2016 9:17 AM
    Subject: Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

    ok, solved the firewall issue. A first test call worked fine. Another one now still gets disconnected after 32s.

    But in FW there are no blocked packets anymore?!

    And I don’t understand why the registration to the same IP and same Port is working, but not later transmission of further SIP packets? that doesn’t sound logical to me. What do you think?

    regards, andre

  • ok, now it is getting weird… actually i don’t see any firewall issues, but i am not able to get a call from outside to my sipgate account. in asterisk nothing is visible, core set verbose is activated. sngrep (on my asterisk server) shows me indeed the invite from sipgate!?

    What I see via sngrep is the following options-flow:
    192.168.3.50:55060 -> 217.10.79.9:5060
    217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK)

    shouldn’t sipgate answer on the same port that the communication initiated??? in this case 55060?

  • Hi, I made some interesting observations regarding this. Remind the following scenario:
    asterisk registers number A towards provider A (sipgate.de)
    asterisk also registers another number towards provider B (tel.t-online.de)

    I make a test call from a remote location, which is registered as well towards provider B.

    What happens on the asterisk?

    have a look below. It seems that the registration to number B of provider B, which sends the incoming call to asterisk, brings asterisk out of order, because the ACKs are sent to the wrong port. after some time the ACK of provider B reaches asterisk on the right port
    (55060), but that is maybe by accident. when the trunk to provider B is disabled it works fine immediately. That explains, why calls to sipgate.de are cancelled after about 30s. But can someone explain that? Is it an issue of provider B (which I
    can’t believe, it is one of the biggest here in germany)?

    217.10.79.9:5060 10.17.46.99:55060
    217.0.23.36:5060 10.17.46.99:65319 10.17.46.99:505
    ──────────┬───────── ──────────┬─────────
    ──────────┬───────── ──────────┬───────── ──────────┬──────
    ▒ │ INVITE (SDP) │
    │ │ │
    ▒ 18:26:47.695820 │ ──────────────────────────>
    │ │ │ │
    ▒ +0.001354 │ 100 Trying │ │
    │ │
    ▒ 18:26:47.697174 │ <────────────────────────── │ │ │ │ ▒ +0.024128 │ 200 OK (SDP) │ │ │ │ ▒ 18:26:47.721302 │ <────────────────────────── │ │ │ │ ▒ +0.418916 │ INVITE (SDP) │ │ │ │ ▒ 18:26:48.140218 │ ────────────────────────>>>
    │ │ │ │
    ▒ +0.000464 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:26:48.140682 │ <<<──────────────────────── │ │ │ │ ▒ +0.047604 │ │ │ ACK │ │ ▒ 18:26:48.188286 │ │ │ ──────────────────────────> │ │
    ▒ +0.033199 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:26:48.221485 │ <<<──────────────────────── │ │ │ │ ▒ +0.413234 │ │ │ ACK │ │ ▒ 18:26:48.634719 │ │ │ ──────────────────────────> │ │
    ▒ +0.073871 │ │ │ ACK
    │ │
    ▒ 18:26:48.708590 │ │ │
    ──────────────────────────> │ │
    ▒ +1.513110 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:26:50.221700 │ <<<──────────────────────── │ │ │ │ ▒ +0.478899 │ │ │ ACK │ │ ▒ 18:26:50.700599 │ │ │ ──────────────────────────> │ │
    ▒ +3.520824 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:26:54.221423 │ <<<──────────────────────── │ │ │ │ ▒ +0.524939 │ │ │ ACK │ │ ▒ 18:26:54.746362 │ │ │ ──────────────────────────> │ │
    ▒ +3.475120 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:26:58.221482 │ <<<──────────────────────── │ │ │ │ ▒ +0.482321 │ │ │ ACK │ │ ▒ 18:26:58.703803 │ │ │ ──────────────────────────> │ │
    ▒ +3.519206 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:27:02.223009 │ <<<──────────────────────── │ │ │ │ ▒ +0.481228 │ │ │ ACK │ │ ▒ 18:27:02.704237 │ │ │ ──────────────────────────> │ │
    ▒ +3.518524 │ 200 OK (SDP) │
    │ │ │
    ▒ 18:27:06.222761 │ <<<──────────────────────── │ │ │ │ │ +0.483722 │ │ │ ACK │ │ │ 18:27:06.706483 │ │ │ ──────────────────────────> │ │
    │ +3.516816 │ 200 OK (SDP) │
    │ │ │
    │ 18:27:10.223299 │ <<<──────────────────────── │ │ │ │ │ +0.484398 │ │ ACK │ │ │ │ 18:27:10.707697 │ │ <────────────────────────── │ │ │ │ +59.790625 │ │ │ BYE │ │ 18:28:10.498322 │ │ │ ────────────────────────────────────────────────────────> │

    Am 15.10.2016 um 15:39 schrieb Andre Gronwald: