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Thema: SIP LINE sign not turning green

  1. #1
    STARFACE Newbie
    Registriert seit
    15.01.2008
    Beiträge
    3

    Standard SIP LINE sign not turning green

    Dear Sirs,

    I cannot get my SIP line to turn green in order to use it to talk to my IP phones
    ( please see attached file for my configuration )

    Could you please send SIP Line config samples so that I may get a green light?

    Appreciating your help and support...
    Miniaturansichten angehängter Grafiken Miniaturansichten angehängter Grafiken sip-line-config.JPG  

  2. #2
    STARFACE Expert
    Benutzerbild von Philipp
    Registriert seit
    08.01.2007
    Ort
    Karlsruhe
    Beiträge
    1.782

    Standard

    Hi carrenoj1, welcome to our forum !

    Your configuration seems to be correct, you didn't made any general errors.
    Are you sure the different "fromdomain" and "host" parameters are correct ???
    Did you configure a port-forwarding in your firewall (port 5060 and 10000-20000 both udp)?

    If yes, we need a sip trace of the registration process. Please connect via SSH on your STARFACE Server. The command für conntecting to asterisk is "asterisk -rvvvvv".

    Then you have to turn on debug mode with "sip debug peer xxx", where "xxx" is the username of your SIP Account. Just paste the complete output here...

    Regards
    Philipp

  3. #3
    STARFACE Newbie
    Registriert seit
    15.01.2008
    Beiträge
    3

    Standard

    Zitat Zitat von Philipp Beitrag anzeigen
    Hi carrenoj1, welcome to our forum !

    Your configuration seems to be correct, you didn't made any general errors.
    Are you sure the different "fromdomain" and "host" parameters are correct ???
    Did you configure a port-forwarding in your firewall (port 5060 and 10000-20000 both udp)?

    If yes, we need a sip trace of the registration process. Please connect via SSH on your STARFACE Server. The command für conntecting to asterisk is "asterisk -rvvvvv".

    Then you have to turn on debug mode with "sip debug peer xxx", where "xxx" is the username of your SIP Account. Just paste the complete output here...

    Regards
    Philipp
    REPLY: -----------------------------------------------------------------------------------------------

    Hello Philipp,

    the output requested follows: thanks...

    -- Registered extension context 'calling'
    -- Added extension '_+.' priority 1 to calling
    -- Added extension '_+.' priority 2 to calling
    -- Added extension '_X.' priority 1 to calling
    -- Added extension '_X.' priority 2 to calling
    -- Added extension '_*.' priority 1 to calling
    -- Added extension '_*.' priority 2 to calling
    -- Added extension '_a.' priority 1 to calling
    -- Added extension '_a.' priority 2 to calling
    -- Added extension 'starfacevm' priority 1 to calling
    -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend)
    == Parsing '/etc/asterisk/cdr_manager.conf': Found
    -- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend)
    == Unregistered 'pgsql' CDR backend
    == Parsing '/etc/asterisk/cdr_pgsql.conf': Found
    -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
    == Parsing '/etc/asterisk/iax.conf': Found
    == Loaded firmware 'iaxy.bin'
    == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
    -- Reloading module 'chan_local.so' (Local Proxy Channel)
    -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
    -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI)
    Reloading SIP
    == Parsing '/etc/asterisk/zapata.conf': Found
    == Parsing '/etc/asterisk/sip.conf': Found
    == Parsing '/etc/asterisk/dusnet.conf': Found
    == Parsing '/etc/asterisk/sip_custom.conf': Found
    -- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decode r)
    -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
    -- Reloading module 'codec_g726.so' (ITU G.726-32kbps G726 Transcoder)
    -- Reloading module 'codec_gsm.so' (GSM/PCM16 (signed linear) Codec Translat or)
    -- Reloading module 'codec_lpc10.so' (LPC10 2.4kbps (signed linear) Voice Co der)
    -- Reloading module 'codec_ulaw.so' (Mu-law Coder/Decoder)
    -- Reloading module 'app_meetme.so' (MeetMe conference bridge)
    == Parsing '/etc/asterisk/meetme.conf': Found
    -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
    == Parsing '/etc/asterisk/voicemail.conf': Found
    == Parsing '/etc/asterisk/sip_notify.conf': Found
    REGISTER 12 headers, 0 lines
    Reliably Transmitting (no NAT) to 10.30.10.4:5060:
    REGISTER sip:10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;rport
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Expires: 600
    Contact: <sip:4434@10.30.10.4>
    Event: registration
    Content-Length: 0


    ---
    12 headers, 3 lines
    Reliably Transmitting (no NAT) to 10.30.10.4:5060:
    NOTIFY sip:PruebaSPHONE@10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;rport
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 108

    Messages-Waiting: yes
    Message-Account: sip:starfacevm@starface.oltpvoice.com.ve
    Voice-Message: 2/0 (0/0)

    ---
    Scheduling destruction of call '55409e270157ad79224fd415190b774a@starface.oltpvo ice.com.ve' in 15000 ms

    <-- SIP read from 10.30.10.4:5060:
    REGISTER sip:10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;rport
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Expires: 600
    Contact: <sip:4434@10.30.10.4>
    Event: registration
    Content-Length: 0


    --- (12 headers 0 lines) ---
    Using latest REGISTER request as basis request
    Sending to 10.30.10.4 : 5060 (NAT)
    [COLOR="Yellow"]Jan 16 17:51:14 ERROR[21061]: chan_sip.c:6802 register_verify: Peer '0002' is trying to register, but not configured as host=dynamic
    Transmitting (NAT) to 10.30.10.4:5060:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;received=10 .30.10.4;rpor t=5060
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>;tag=as2c0706eb
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    ---
    Scheduling destruction of call '132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4' in 15000 ms

    <-- SIP read from 10.30.10.4:5060:
    NOTIFY sip:PruebaSPHONE@10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;rport
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 108

    Messages-Waiting: yes
    Message-Account: sip:starfacevm@starface.oltpvoice.com.ve
    Voice-Message: 2/0 (0/0)

    --- (12 headers 3 lines) ---
    Transmitting (no NAT) to 10.30.10.4:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;received=10 .30.10.4;rpor t=5060
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>;tag=as481f29e8
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Content-Length: 0


    ---

    <-- SIP read from 10.30.10.4:5060:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;received=10 .30.10.4;rpor t=5060
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>;tag=as2c0706eb
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    --- (9 headers 0 lines) ---
    Destroying call '132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4'

    <-- SIP read from 10.30.10.4:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;received=10 .30.10.4;rpor t=5060
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>;tag=as481f29e8
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Content-Length: 0


    --- (10 headers 0 lines) ---
    Destroying call '55409e270157ad79224fd415190b774a@starface.oltpvoi ce.com.ve'
    starface*CLI>

  4. #4
    STARFACE Newbie
    Registriert seit
    15.01.2008
    Beiträge
    3

    Standard

    REPLY: -----------------------------------------------------------------------------------------------

    Hi Philipp,

    When trying to register my X-LITE softphone, this is what I get:
    starface*CLI> sip debug peer 0002
    SIP Debugging Enabled for IP: 10.30.10.4:5060
    Jan 16 18:07:07 ERROR[21061]: chan_sip.c:6802 register_verify: Peer 'XLITE' is trying to register, but not configured as host=dynamic
    starface*CLI> starface*CLI> sip debug peer 0002
    starface*CLI> SIP Debugging Enabled for IP: 10.30.10.4:5060
    starface*CLI> Jan 16 18:07:07 ERROR[21061]: chan_sip.c:6802 register_verify: Peer 'XLITE' is trying to register, but not configured as host=dynamic
    starface*CLI> starface*CLI>

    Thanks,



    Zitat Zitat von carrenoj1 Beitrag anzeigen
    REPLY: -----------------------------------------------------------------------------------------------

    Hello Philipp,

    the output requested follows: thanks...

    -- Registered extension context 'calling'
    -- Added extension '_+.' priority 1 to calling
    -- Added extension '_+.' priority 2 to calling
    -- Added extension '_X.' priority 1 to calling
    -- Added extension '_X.' priority 2 to calling
    -- Added extension '_*.' priority 1 to calling
    -- Added extension '_*.' priority 2 to calling
    -- Added extension '_a.' priority 1 to calling
    -- Added extension '_a.' priority 2 to calling
    -- Added extension 'starfacevm' priority 1 to calling
    -- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend)
    == Parsing '/etc/asterisk/cdr_manager.conf': Found
    -- Reloading module 'cdr_pgsql.so' (PostgreSQL CDR Backend)
    == Unregistered 'pgsql' CDR backend
    == Parsing '/etc/asterisk/cdr_pgsql.conf': Found
    -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2))
    == Parsing '/etc/asterisk/iax.conf': Found
    == Loaded firmware 'iaxy.bin'
    == Parsing '/etc/asterisk/iaxprov.conf': Found
    -- Loaded provisioning template 'default'
    -- Reloading module 'chan_local.so' (Local Proxy Channel)
    -- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
    -- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI)
    Reloading SIP
    == Parsing '/etc/asterisk/zapata.conf': Found
    == Parsing '/etc/asterisk/sip.conf': Found
    == Parsing '/etc/asterisk/dusnet.conf': Found
    == Parsing '/etc/asterisk/sip_custom.conf': Found
    -- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decode r)
    -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
    -- Reloading module 'codec_g726.so' (ITU G.726-32kbps G726 Transcoder)
    -- Reloading module 'codec_gsm.so' (GSM/PCM16 (signed linear) Codec Translat or)
    -- Reloading module 'codec_lpc10.so' (LPC10 2.4kbps (signed linear) Voice Co der)
    -- Reloading module 'codec_ulaw.so' (Mu-law Coder/Decoder)
    -- Reloading module 'app_meetme.so' (MeetMe conference bridge)
    == Parsing '/etc/asterisk/meetme.conf': Found
    -- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
    == Parsing '/etc/asterisk/voicemail.conf': Found
    == Parsing '/etc/asterisk/sip_notify.conf': Found
    REGISTER 12 headers, 0 lines
    Reliably Transmitting (no NAT) to 10.30.10.4:5060:
    REGISTER sip:10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;rport
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Expires: 600
    Contact: <sip:4434@10.30.10.4>
    Event: registration
    Content-Length: 0


    ---
    12 headers, 3 lines
    Reliably Transmitting (no NAT) to 10.30.10.4:5060:
    NOTIFY sip:PruebaSPHONE@10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;rport
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 108

    Messages-Waiting: yes
    Message-Account: sip:starfacevm@starface.oltpvoice.com.ve
    Voice-Message: 2/0 (0/0)

    ---
    Scheduling destruction of call '55409e270157ad79224fd415190b774a@starface.oltpvo ice.com.ve' in 15000 ms

    <-- SIP read from 10.30.10.4:5060:
    REGISTER sip:10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;rport
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Expires: 600
    Contact: <sip:4434@10.30.10.4>
    Event: registration
    Content-Length: 0


    --- (12 headers 0 lines) ---
    Using latest REGISTER request as basis request
    Sending to 10.30.10.4 : 5060 (NAT)
    [COLOR="Yellow"]Jan 16 17:51:14 ERROR[21061]: chan_sip.c:6802 register_verify: Peer '0002' is trying to register, but not configured as host=dynamic
    Transmitting (NAT) to 10.30.10.4:5060:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;received=10 .30.10.4;rpor t=5060
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>;tag=as2c0706eb
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    ---
    Scheduling destruction of call '132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4' in 15000 ms

    <-- SIP read from 10.30.10.4:5060:
    NOTIFY sip:PruebaSPHONE@10.30.10.4 SIP/2.0
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;rport
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Max-Forwards: 70
    Event: message-summary
    Content-Type: application/simple-message-summary
    Content-Length: 108

    Messages-Waiting: yes
    Message-Account: sip:starfacevm@starface.oltpvoice.com.ve
    Voice-Message: 2/0 (0/0)

    --- (12 headers 3 lines) ---
    Transmitting (no NAT) to 10.30.10.4:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;received=10 .30.10.4;rpor t=5060
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>;tag=as481f29e8
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Content-Length: 0


    ---

    <-- SIP read from 10.30.10.4:5060:
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK6c45fbf7;received=10 .30.10.4;rpor t=5060
    From: <sip:0002@10.30.10.4>;tag=as2c0706eb
    To: <sip:0002@10.30.10.4>;tag=as2c0706eb
    Call-ID: 132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4
    CSeq: 102 REGISTER
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Content-Length: 0


    --- (9 headers 0 lines) ---
    Destroying call '132c67f96b03c4ce4ab8259d7d2ade55@10.30.10.4'

    <-- SIP read from 10.30.10.4:5060:
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.30.10.4:5060;branch=z9hG4bK46e37c07;received=10 .30.10.4;rpor t=5060
    From: "STARFACE" <sip:PruebaSPHONE@starface.oltpvoice.com.ve>;tag=a s481f29e8
    To: <sip:PruebaSPHONE@10.30.10.4>;tag=as481f29e8
    Call-ID: 55409e270157ad79224fd415190b774a@sta...voic e.com.ve
    CSeq: 102 NOTIFY
    User-Agent: STARFACE PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Contact: <sip:PruebaSPHONE@10.30.10.4>
    Content-Length: 0


    --- (10 headers 0 lines) ---
    Destroying call '55409e270157ad79224fd415190b774a@starface.oltpvoi ce.com.ve'
    starface*CLI>

  5. #5
    STARFACE Expert
    Benutzerbild von Philipp
    Registriert seit
    08.01.2007
    Ort
    Karlsruhe
    Beiträge
    1.782

    Standard

    Hi,

    thx for the sip trace.

    I guess the two parameteres "fromdomain" and "host" are incorrect because these two adresses are from a private network (more information) which will not be routed through the internet.

    Did you get these parameters from your provider ? What provider do you use?

    Another question: Is your username for this SIP account really "0002" ??? Normally user names consists of more than 4 digits, did you probably mixed it up with your STARFACE LoginID ???

    starface*CLI> Jan 16 18:07:07 ERROR[21061]: chan_sip.c:6802 register_verify: Peer 'XLITE' is trying to register, but not configured as host=dynamic
    How did you configured xlite in STARACE? Which profile did you use? "Standard SIP" would be the correct profile...

    Regards
    Philipp

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