Posts by zmagyar

  • SF Tot

    Hallo Jochen,


    [root@starface ~]# df -h
    Dateisystem Größe Benut Verf Ben% Eingehängt auf
    /dev/sda2 7,2G 4,6G 2,2G 68% /
    /dev/sda1 100M 20M 75M 21% /boot
    /dev/shm 501M 0 501M 0% /dev/shm


    stimmt das?


    Danke für deine Mühe,
    Zsolt

  • SF Tot

    Hallo epg789, danke für die Hinweis. Ich vermute dass Du meinst dass die Lösung sollte "Der Starface Support schaltete sich auf und hat den TomCat Webserver neu heruntergeladen und installiert, dann ging das wieder." sein?
    Ich bin nicht überzeugt, weil im Logs klar steht dass er sich mit dem DB nicht verbinden kann. Ich vermute es liegt am beschädigte DB, aber ich habe keine Ahnung wie das zu behoben. Wie geschrieben, ich habe auch das reset Skripte ausgeführt, was eigentlich alles neu eingerichtet sollte, hat es aber nicht gemacht.
    Ich denke wir sollten es neu installieren, und danach eine Rücksicherung machen. Ich würde aber gerne vom Support hören ob das der richtige Weg ist, und wo kann ich Tipps finden wie die Neuinstallation man durchführen soll.
    Danke im Voraus,
    Zsolt

  • SF Tot

    Guten Morgen,
    leider ist uns Heute nicht ganz gut gegangen.
    Erst versuchte ich eine Nummer anzurufen, und nach es zweimal geklingelt hat, hat mein iSoftPhone ist mir "internal server error" gegeben.
    Da mir schon mehrere Email über neue Updates gekommen sind, habe ich entscheidet den Server upgraden.


    Das upgrade war irgendwie ungewöhnliche langsam, aber nach download, backup das update datebase (Schritt 3.) hat sich abgebrochen mit der Fehlermeldung dass es sich auf letzte Stand wiederherstellen wird. Da aber nach einige Minuten nichts passiert, habe ich mich eingelogt durch ssh. Das geht OK, und astersik war auch da.


    Web Front end lauft auber nicht, mit Fehlermeldung:
    HTTP Status 404 -
    type Status report
    message
    description The requested resource () is not available.


    Dann habe ich hier im Forum angeschaut
    Versuchte:
    [root@starface ~]# /etc/init.d/postgresql restart
    postgresql-Dienst stoppen: [ OK ]
    postgresql-Dienst starten: [ OK ]
    [root@starface ~]#


    [root@starface ~]# /etc/init.d/tomcat5 restart
    Using CATALINA_BASE: /opt/tomcat
    Using CATALINA_HOME: /opt/tomcat
    Using CATALINA_TMPDIR: /opt/tomcat/temp
    Using JRE_HOME: /usr/lib/jvm/java-sun
    waiting for processes to exit
    Using CATALINA_BASE: /opt/tomcat
    Using CATALINA_HOME: /opt/tomcat
    Using CATALINA_TMPDIR: /opt/tomcat/temp
    Using JRE_HOME: /usr/lib/jvm/java-sun
    3876: Alte Priorität: 0, neue Priorität: 0
    [root@starface ~]#


    Danach habe ich die Logs angeschaut, und dort steht:
    [2011-10-30 03:02:47,216] ERROR de.vertico.starface.update.UpdateController --- preloading autoErrorGenerate-file
    [2011-10-31 03:02:50,061] ERROR de.vertico.starface.update.UpdateController --- preloading autoErrorGenerate-file
    [2011-10-31 07:38:45,691] ERROR de.vertico.starface.update.UpdateController --- preloading autoErrorGenerate-file
    [2011-10-31 07:56:00,813] ERROR de.vertico.starface.StarfaceDataSource getConnection: Unable to get pooled db-connection
    [2011-10-31 07:56:00,814] ERROR de.vertico.starface.CATManager getConnection failed: No db connection avaiable
    [2011-10-31 07:56:00,819] ERROR de.vertico.starface.persistence.hibernate.HibernateConnector Initial SessionFactory creation failed.
    java.lang.RuntimeException: No db connection available
    at de.vertico.starface.CATManager.getConnection(CATManager.java:739)
    at de.vertico.starface.persistence.hibernate.StarfaceConnectionProvider.getConnection(StarfaceConnectionProvider.java:44)
    at org.hibernate.cfg.SettingsFactory.buildSettings(SettingsFactory.java:76)
    at org.hibernate.cfg.Configuration.buildSettings(Configuration.java:1933)
    at org.hibernate.cfg.Configuration.buildSessionFactory(Configuration.java:1216)
    at de.vertico.starface.persistence.hibernate.HibernateConnector.init(HibernateConnector.java:57)
    at de.vertico.starface.persistence.hibernate.HibernateConnector.init(HibernateConnector.java:48)
    at de.vertico.starface.CATManager.initFull(CATManager.java:243)
    at de.vertico.starface.StarfaceStarter.contextInitializedImpl(StarfaceStarter.java:257)
    at de.vertico.starface.StarfaceStarter$1.run(StarfaceStarter.java:131)
    [2011-10-31 07:56:00,847] FATAL de.vertico.starface.StarfaceStarter Could not start Application. Reason:
    java.lang.ExceptionInInitializerError
    at de.vertico.starface.persistence.hibernate.HibernateConnector.init(HibernateConnector.java:63)
    at de.vertico.starface.persistence.hibernate.HibernateConnector.init(HibernateConnector.java:48)
    at de.vertico.starface.CATManager.initFull(CATManager.java:243)
    at de.vertico.starface.StarfaceStarter.contextInitializedImpl(StarfaceStarter.java:257)
    at de.vertico.starface.StarfaceStarter$1.run(StarfaceStarter.java:131)
    Caused by: java.lang.RuntimeException: No db connection available
    at de.vertico.starface.CATManager.getConnection(CATManager.java:739)
    at de.vertico.starface.persistence.hibernate.StarfaceConnectionProvider.getConnection(StarfaceConnectionProvider.java:44)
    at org.hibernate.cfg.SettingsFactory.buildSettings(SettingsFactory.java:76)
    at org.hibernate.cfg.Configuration.buildSettings(Configuration.java:1933)
    at org.hibernate.cfg.Configuration.buildSessionFactory(Configuration.java:1216)
    at de.vertico.starface.persistence.hibernate.HibernateConnector.init(HibernateConnector.java:57)
    ... 4 more
    [2011-10-31 07:56:00,890] ERROR org.apache.catalina.core.ContainerBase.[Catalina].[localhost].[/] Exception sending context initialized event to listener instance of class de.vertico.starface.StarfaceStarter
    java.lang.RuntimeException: java.lang.ExceptionInInitializerError
    at de.vertico.starface.StarfaceStarter.contextInitialized(StarfaceStarter.java:191)
    at org.apache.catalina.core.StandardContext.listenerStart(StandardContext.java:3764)
    at org.apache.catalina.core.StandardContext.start(StandardContext.java:4216)
    at org.apache.catalina.core.ContainerBase.start(ContainerBase.java:1014)
    at org.apache.catalina.core.StandardHost.start(StandardHost.java:736)


    Es scheint etwas mit java nicht stimmt, und die Verbindung zu DB geht nicht.


    Ich habe auch ein reset gemacht laut http://support.starface.de/for…fohlen-!!&p=1491#post1491
    hat aber nicht geholfen....


    Ich würde mich ehrlich für eine schnelle Hilfe bedanken...


    Zsolt

  • Fehler 411

    no, I do not know how to activate debug, pls write me the command


    to sipgate support: do you know what is the best way to get support from them, at the time we were signing contract with them, I asked if we can get support and the reply was very arrogant, saying "if there are problems then check your equipment" :-), meaning I can't get any support from them. Otherwise they are reliable and good, but the support is nonexsistent.


    Thanks
    Zsolt

  • Fehler 411

    Hi Phillip, I'm glad to hear you again.
    This is step forward, so the announcement comes from provider, but the reason for it should be in the starface.
    I rebooted the starface appliance, maybe that helps.
    Do you want to check the logs (I'm not sure if they can contain something which can help you now), or we wait if and when it happens again and then act?
    The trouble is that it seems it is intermittent, you try to call 5-6 times and then it suddenly works fine, once the connection is established then all is OK. So if we will have to debug live then it is a question if we will be able to reproduce it.
    It would be better if the logs can show this somehow.
    Rgds


    Zsolt

  • Fehler 411

    Dear Support, since yesterday we are getting ocasional errors of 411.


    When we dial we get a german mail voice reply "es ist ein Fehler aufgetretten, fehler 411"


    Now I'm not sure if this is a message from the starface appliance or from the sipgate VoIP provider.


    I found some explanation on the web for the error http://books.google.com/books?…Length%20required&f=false


    but I have no idea what does that mean, and where to look for the fix.
    Yesterday we had unusually many lines open (7 I think) so I thought it is some kind of limitation, but today it happened with 4 lines open while we tried to dial the 5th.


    Pls help
    Thanks
    Zsolt

  • Caller Number not visible when call forward is active

    Hi Torsten,


    I guess this will end up as feature request, but let me clarify what I'm aiming to:


    as confirmed in your reply to first topic, if a call forward in SF is activated, SF does send the calling phone number to the phone call is forwarded to, however, due to limitation of our (an maybe other providers) this call number actually does not arrive to the phone call is forwarded to.
    Consequently, if I miss the call on the phone, the call if forwarded to, I will only see the missed call, but I will not see who was calling.


    Now, to overcome this limitation of our provider, it would be nice if the forwarded call would show up in the call list of the user, in the SF, simply as call which was forwarded, together with the call number of the party who was calling. In this way I would have a chance to figure out who was calling.


    In current situation the calling number is known to SF but I have no way to find it out, and it is lost forever (unless I go and dig into the logs, I assume)


    So, the bottom line, is possible I would file a feature request:
    The call list should show, beside the incoming, outgoing, missed, also the forwarded calls.


    Thanks
    Zsolt

  • [Solved] unable to return to call in case trying to transfer to nonexsisting phone

    This one is a bit specific and I just want to know if you are aware of it:


    We get an incoming call, this get picked up by first person.
    Now this first person wants to transfer this call to a second person.
    So the first person puts the call on hold, dials the phone of the second person, but the SIP client of the second person phone is not started so the first person gets "service unavailable".


    So far so good.


    Now the first person wants to return to the call he put on hold, but can't.
    This is the problem.


    Is this expected?


    Thanks
    Zsolt

  • registration error: 404 - Not found

    Dear Support


    When I start my eyeBeam SIP client, I get an error message "registration error: 404 - Not found", however, all is working fine.


    Now some extra info, do not get confused!


    Recently I got this error when I assigned to my user another phone and restricted that phone to a fix IP address and this phone was not present on the net.
    Once I removed the IP restriction, the problem went away.


    However, now I got it again, to try to fix it I completely removed the second phone but the problem remained. The primary phone is not restricted to IP.


    Any idea?
    Which output from console do you require to troubleshoot this further?


    Rgds
    Zsolt

  • how to configure incoming VoIP call

    I works fine even without the manual config. obviously the problem was that the phone number bound to sipgate line was nowhere defined. Now you learned again something new :-).


    Reward me with one more answer please :)
    OK, now we have one line and this line has a number bound to it, and I assigned the number to myself and all is fine. BUT, let's say I want now inbound number for each our employee (10).


    In theory I could open 10 accounts at sipgate, each would get a number. I would define the number for each line and then assign it to user. And as long as we are logged on with all 10 accounts the incoming calls would ring at the right place, correct?


    Or are you aware of any other way to get 10 inbound numbers (with one sipgate account only)?


    Reason I'm asking this is because at sipgate you have free account but with higher calling rates, and accounts with monthly fee but with lower calling rates. So if I want the low calling rates I need to create 10 accounts and pay the monthly fee for each.


    Thanks
    Zsolt

  • how to configure incoming VoIP call

    Hi Phillip, I did the other part, I configured the single number as you said in the line configuration and assigned it to my user and now it works :)
    Whether the extended config you gave me is needed or not, I don't know. If you are interested let me know, if not, we leave it as it is, don't touch the running system :)


    Maybe we hear us soon as I was promised that you will help me with the appliance config.
    Have a nice weekend
    Zsolt

  • how to configure incoming VoIP call

    It is hard to do it right if you do not know what are you doing (this is considering the wrong case in the channel number), I just copy and paste and pray :)


    Here is the new trace


    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:


    --- (0 headers 0 lines) Nat keepalive ---
    == Primary D-Channel on span 3 down
    == Primary D-Channel on span 4 down
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:
    INVITE sip:7@78.2.107.205 SIP/2.0
    Record-Route: <sip:217.10.79.9;lr=on;ftag=as29954471>
    Record-Route: <sip:172.20.40.4;lr=on>
    Record-Route: <sip:217.10.79.9;lr=on;ftag=as29954471>
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7a61.ea732656.0
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7a61.ea732656.0
    Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK09a79071
    Via: SIP/2.0/UDP 217.10.67.135:5060;branch=z9hG4bK09a79071;rport=5060
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as29954471
    To: <sip:00498105399036@sipgate.de>
    Contact: <sip:0038598369691@217.10.67.135>
    Call-ID: 3f380b8e2029255b4ec0be634783adc1@sipgate.de
    CSeq: 102 INVITE
    Max-Forwards: 67
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 449


    v=0
    o=root 25834 25834 IN IP4 217.10.67.135
    s=session
    c=IN IP4 217.10.77.21
    t=0 0
    m=audio 37124 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=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=direction:active
    a=nortpproxy:yes


    --- (18 headers 21 lines) ---
    Using INVITE request as basis request - 3f380b8e2029255b4ec0be634783adc1@sipgate.de
    Sending to 217.10.79.9 : 5060 (NAT)
    Found peer '2370593'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 112
    Found RTP audio format 101
    Peer audio RTP is at port 217.10.77.21:37124
    Peer video RTP is at port 217.10.77.21:65535
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format telephone-event
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Looking for 7 in sipgate-incoming (domain 78.2.107.205)
    Reliably Transmitting (NAT) to 217.10.79.9:5060:
    SIP/2.0 484 Address Incomplete
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7a61.ea732656.0;received=217.10.79.9
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7a61.ea732656.0
    Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK09a79071
    Via: SIP/2.0/UDP 217.10.67.135:5060;branch=z9hG4bK09a79071;rport=5060
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as29954471
    To: <sip:00498105399036@sipgate.de>;tag=as0bb79a2b
    Call-ID: 3f380b8e2029255b4ec0be634783adc1@sipgate.de
    CSeq: 102 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0



    ---
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:
    ACK sip:7@78.2.107.205 SIP/2.0
    Max-Forwards: 10
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7a61.ea732656.0
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7a61.ea732656.0
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as29954471
    Call-ID: 3f380b8e2029255b4ec0be634783adc1@sipgate.de
    To: <sip:00498105399036@sipgate.de>;tag=as0bb79a2b
    CSeq: 102 ACK
    Content-Length: 0
    X-hint: rr-enforced



    --- (10 headers 0 lines) ---
    Destroying call '3f380b8e2029255b4ec0be634783adc1@sipgate.de'
    == Primary D-Channel on span 3 down
    == Primary D-Channel on span 4 down
    == Primary D-Channel on span 3 down
    starface*CLI> e
    <-- SIP read from 217.10.79.9:5060:


    --- (0 headers 0 lines) Nat keepalive ---
    == Primary D-Channel on span 4 down
    starface*CLI> exit


    ignore the D-Channel down messages, our ISDN cards just died...the SF appliance is already on the way to us :)


    Thanks for help
    Zsolt

  • how to configure incoming VoIP call

    I pasted, it (fist removed what was alreday there and this was
    [sipgate-incoming]
    exten => _X.,1,Set(channelname=sipgate-incoming)
    exten => _X.,2,Goto(incoming,${EXTEN},1)


    exten => _+X.,1,Set(channelname=sipgate-incoming)
    exten => _+X.,2,Goto(incoming,${EXTEN:1},1)


    so now it looks like:
    [sipgate-incoming]
    exten => _X.,1,Set(channelname=Sipgate-incoming)
    exten => _X.,2,Set(var_to=${SIP_HEADER(To)})
    exten => _X.,3,Set(firstcut=${CUT(var_to|:|2)})
    exten => _X.,4,Set(secoundcut=${CUT(firstcut|@|1)})
    exten => _X.,5,Goto(incoming,${secoundcut},1)
    exten => _X.,6,Hangup


    but it does not work...


    However, before I send you another trace, some clarifications, as some parameters has to do with number 7 and I do not know what are they for, so let me just list them and you tell me if something is wrong:


    we have 7 lines configured in starface, two are ISDN, and 5 are (VoIP) providers
    the sipgate line is number 5
    The line prefix of this line, in the extended tab of the line configuration is 7, i.e it is **7*
    The number area tab of this line, is Number Type: single number, Number:00385(51)7
    The 00385(51) the SF filled out by itself but the 7 I wrote in. It requested me to write in some number, and the other lines have other (one digit) number there. This number has to be different then the number in the other lines, so I put 7, although I have no idea what are these for.


    So this is about "seven" in our config. Is that OK?


    Thanks for help so far
    Zsolt

  • how to configure incoming VoIP call

    Hi Phillip, you were right, with debug some sign of life is visible...here is the log


    starface*CLI> sip debug peer 2370593
    SIP Debugging Enabled for IP: 217.10.79.9:5060
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:


    --- (0 headers 0 lines) Nat keepalive ---
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:


    --- (0 headers 0 lines) Nat keepalive ---
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:
    INVITE sip:7@78.2.107.205 SIP/2.0
    Record-Route: <sip:217.10.79.9;lr=on;ftag=as3fab6c6e>
    Record-Route: <sip:172.20.40.4;lr=on>
    Record-Route: <sip:217.10.79.9;lr=on;ftag=as3fab6c6e>
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7276.79bc7c91.0
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7276.79bc7c91.0
    Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK3988ca4c
    Via: SIP/2.0/UDP 217.10.67.4:5060;branch=z9hG4bK3988ca4c;rport=5060
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as3fab6c6e
    To: <sip:00498105399036@sipgate.de>
    Contact: <sip:0038598369691@217.10.67.4>
    Call-ID: 0b0177946207e46317ea3f0d7595a321@sipgate.de
    CSeq: 102 INVITE
    Max-Forwards: 67
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 447


    v=0
    o=root 10649 10649 IN IP4 217.10.67.4
    s=session
    c=IN IP4 217.10.77.24
    t=0 0
    m=audio 47598 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=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    a=direction:active
    a=nortpproxy:yes


    --- (18 headers 21 lines) ---
    Using INVITE request as basis request - 0b0177946207e46317ea3f0d7595a321@sipgate.de
    Sending to 217.10.79.9 : 5060 (NAT)
    Found peer '2370593'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 97
    Found RTP audio format 18
    Found RTP audio format 112
    Found RTP audio format 101
    Peer audio RTP is at port 217.10.77.24:47598
    Peer video RTP is at port 217.10.77.24:65535
    Found description format PCMA
    Found description format PCMU
    Found description format GSM
    Found description format iLBC
    Found description format G729
    Found description format G726-32
    Found description format telephone-event
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x51e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Looking for 7 in sipgate-incoming (domain 78.2.107.205)
    Reliably Transmitting (NAT) to 217.10.79.9:5060:
    SIP/2.0 484 Address Incomplete
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7276.79bc7c91.0;received=217.10.79.9
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7276.79bc7c91.0
    Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK3988ca4c
    Via: SIP/2.0/UDP 217.10.67.4:5060;branch=z9hG4bK3988ca4c;rport=5060
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as3fab6c6e
    To: <sip:00498105399036@sipgate.de>;tag=as5cccd217
    Call-ID: 0b0177946207e46317ea3f0d7595a321@sipgate.de
    CSeq: 102 INVITE
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0



    ---
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:
    ACK sip:7@78.2.107.205 SIP/2.0
    Max-Forwards: 10
    Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK7276.79bc7c91.0
    Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK7276.79bc7c91.0
    From: "0038598369691" <sip:0038598369691@sipgate.de>;tag=as3fab6c6e
    Call-ID: 0b0177946207e46317ea3f0d7595a321@sipgate.de
    To: <sip:00498105399036@sipgate.de>;tag=as5cccd217
    CSeq: 102 ACK
    Content-Length: 0
    X-hint: rr-enforced



    --- (10 headers 0 lines) ---
    Destroying call '0b0177946207e46317ea3f0d7595a321@sipgate.de'
    starface*CLI>
    <-- SIP read from 217.10.79.9:5060:


    --- (0 headers 0 lines) Nat keepalive ---
    starface*CLI> exit


    Any idea?
    Thanks
    Zsolt