SIP Client Trennung nach 6s und keine Ton Übertragung

  • Moin Moin,


    ich habe ein merkwürdiges Problem, ich nutze den Bria SIP Client auch zum testen ein SNOM Telefon.


    Schilderung:
    - Ich kann jemanden Anrufen, ich höre mein Gesprächspartner. Er leider mich nicht.
    - Mein Telefon klingelt nicht, wenn ich angerufen werde.
    - Nach ca. 6s wird der Anruf von der Anlage getrennt.


    Ich habe folgende Ports auf der Firewall freigeben:
    5060,5061,5222,3090,50080,50081,4569/tcp_udp
    10000-20000,20000-65535/udp


    Habt ihr eine Idee?



    Logfile (/var/log/asterisk/full):
    [Feb 1 09:42:24] VERBOSE[10559][C-00000086] netsock2.c: == Using SIP RTP TOS bits 184
    [Feb 1 09:42:24] VERBOSE[10559][C-00000086] netsock2.c: == Using SIP RTP CoS mark 5
    [Feb 1 09:42:24] WARNING[10559][C-00000086] chan_sip.c: Got Opus useinbandfec=1
    [Feb 1 09:42:24] WARNING[10559][C-00000086] chan_sip.c: Got Opus useinbandfec=1
    [Feb 1 09:42:24] VERBOSE[10559][C-00000086] res_rtp_asterisk.c: > 0x7fc974bd1440 -- Strict RTP learning after remote address set to: 80.187.102.95:64164
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] pbx.c: -- Executing [0441390101024@dialstart:1] AGI("SIP/sip.nmeyer-00000045", "agi:async,,,,") in new stack
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (X-UCI_CALLID)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (SIPAddHeader) Options: (X-UCI_CALLID:f70b5775-8cf0-436a-a2e9-ebfd7a170303)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(num,i)=0441390101024)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(num-pres,i)=allowed)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(name,i)=Paul Folkers)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(name-pres)=allowed)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (Call-Info)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (SIPAddHeader) Options: (Call-Info:<http://192.168.16.10:50080/ava…6dcee0e1c62e98d9c70eb6e1d>;purpose=icon)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=Nils Meyer)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(number)=23)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] res_agi.c: -- AGI Script Executing Application: (Dial) Options: (SIP/sip.pf&SIP/buero5.1.snom720,10,wWtT)
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] netsock2.c: == Using SIP RTP TOS bits 184
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] netsock2.c: == Using SIP RTP CoS mark 5
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] netsock2.c: == Using SIP RTP TOS bits 184
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] netsock2.c: == Using SIP RTP CoS mark 5
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] app_dial.c: -- Called SIP/sip.pf
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] app_dial.c: -- Called SIP/buero5.1.snom720
    [Feb 1 09:42:24] WARNING[10330] chan_sip.c: sip_xmit of 0x7fc9ac034e60 (len 763) to 192.168.16.156:55690 returned -2: Success
    [Feb 1 09:42:24] ERROR[10330] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] app_dial.c: -- SIP/sip.pf-00000046 is ringing
    [Feb 1 09:42:24] VERBOSE[13375][C-00000086] app_dial.c: -- SIP/buero5.1.snom720-00000047 is ringing
    [Feb 1 09:42:25] VERBOSE[13375][C-00000086] res_rtp_asterisk.c: > 0x7fc97812d460 -- Strict RTP learning after remote address set to: 192.168.16.140:11728
    [Feb 1 09:42:25] VERBOSE[11000][C-00000086] res_rtp_asterisk.c: > 0x7fc97812d460 -- Strict RTP learning after remote address set to: 192.168.16.140:11728
    [Feb 1 09:42:25] VERBOSE[13375][C-00000086] app_dial.c: -- SIP/buero5.1.snom720-00000047 answered SIP/sip.nmeyer-00000045
    [Feb 1 09:42:25] VERBOSE[13375][C-00000086] res_rtp_asterisk.c: > 0x7fc97812d460 -- Strict RTP switching to RTP remote address 192.168.16.140:11728 as source
    [Feb 1 09:42:26] VERBOSE[13375][C-00000086] res_rtp_asterisk.c: > 0x7fc97812d460 -- Strict RTP learning complete - Locking on source address 192.168.16.140:11728
    [Feb 1 09:42:37] WARNING[10559] chan_sip.c: Retransmission timeout reached on transmission 143678_rel51ZDcxMGIzZDdiY2Y2MTAwOGM3ZDZmMmE3MTU5NTQ1NzI for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki…y/AST/SIP+Retransmissions
    Packet timed out after 12096ms with no response
    [Feb 1 09:42:37] WARNING[10559] chan_sip.c: Hanging up call 143678_rel51ZDcxMGIzZDdiY2Y2MTAwOGM3ZDZmMmE3MTU5NTQ1NzI - no reply to our critical packet (see https://wiki.asterisk.org/wiki…y/AST/SIP+Retransmissions).
    [Feb 1 09:42:37] VERBOSE[13375][C-00000086] pbx.c: -- Executing [h@dialstart:1] NoOp("SIP/sip.nmeyer-00000045", "HC 18") in new stack
    [Feb 1 09:42:37] VERBOSE[13375][C-00000086] pbx.c: -- Executing [h@dialstart:2] Goto("SIP/sip.nmeyer-00000045", "_exit_,0") in new stack
    [Feb 1 09:42:37] VERBOSE[13375][C-00000086] pbx.c: -- Goto (dialstart,_exit_,0)
    [Feb 1 09:42:37] VERBOSE[13375][C-00000086] pbx.c: == Spawn extension (dialstart, 0441390101024, 1) exited non-zero on 'SIP/sip.nmeyer-00000045'
    [Feb 1 09:42:37] NOTICE[11000][C-00000086] chan_sip.c: RTP stats from peer <sip:buero5.1.snom720@192.168.16.140:53384;transport=Tls;line=2rmrq5a3>: Rx/Total_Rx_Pkts=0,Rx-Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 Tx/Total_Tx_Pkts=603,Tx_Pkts=603,Remote_Tx_Pkts=603, CallID: 2112050b0f97eb486461e5cd1edb954f@192.168.16.10:5061, SipAddHeader01: <none>

Jetzt mitmachen!

Sie haben noch kein Benutzerkonto auf unserer Seite? Registrieren Sie sich kostenlos und nehmen Sie an unserer Community teil!