Hallo zusammen,
ein Kunde von uns hat derzeit Probleme mit dem Telekom SIP-Trunk, welcher direkt in der Starface Regsitriert wird.
Das Profil wurde (aufgrund der Probleme und zur weiteren Analyse) wie folgt angepasst:
transport: TCP und encryption: no
Problembeschreibung:
Ein Telefonat wird gestartet und verläuft auch die ersten 30 Sekunden ohne Probleme, dann ist der externe Gesprächspartner plötzlich nicht mehr zu hören.
Die Teilnehmer sind folgende (wie auch in den Logs ersichtlich:
Rufnummer extern: +4971411234311 (anegerufener Teilnehmer)
Rufnummer auf der Anlage: +49408912347686 (initiierender Teilnehmer auf der Starface) (Softphone: Softphone.PT)
Das Netzwerk ist wie folgt aufgebaut:
Zum Trunk gehöriger Telekom Anschluss <---> LANCOM 1783VA (Portweiterleitung aller VoIP relevanten Ports auf die Sophos) <---> Sophos (Portweiterleitung aller VoIP relevanten Ports auf die STARFACE) <---> Starface Compact
Die Sophos ist Multi-WAN angebungen, jedoch wird der gesamte Traffic der Starface per Multipath Rule über die Telekom Leitung geschickt. Alles andere über die zweite Leitung (Vodafone Kabel)
PBX-Log:
[Mar 16 09:45:18] VERBOSE[28120][C-00001f80] netsock2.c: == Using SIP VIDEO TOS bits 136
[Mar 16 09:45:18] VERBOSE[28120][C-00001f80] netsock2.c: == Using SIP VIDEO CoS mark 6
[Mar 16 09:45:18] VERBOSE[28120][C-00001f80] netsock2.c: == Using SIP RTP TOS bits 184
[Mar 16 09:45:18] VERBOSE[28120][C-00001f80] netsock2.c: == Using SIP RTP CoS mark 5
[Mar 16 09:45:18] VERBOSE[24042][C-00001f80] res_rtp_asterisk.c: > 0x7fd8548e44b0 -- Strict RTP learning after remote address set to: 192.168.77.146:4004
[Mar 16 09:45:18] VERBOSE[28120][C-00001f80] pbx.c: > Channel SIP/Softphone.PT-0000107d was answered
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] pbx.c: -- Executing [+4971411234311@dialstart:1] AGI("SIP/Softphone.PT-0000107d", "agi:async,X-UCI_ORIGINATE: dcbbaa8a-558c-4a3e-955c-1acc29a1549b,,,") in new stack
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_rtp_asterisk.c: > 0x7fd8548e44b0 -- Strict RTP switching to RTP remote address 192.168.77.146:4004 as source
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (X-UCI_ORIGINATE)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (X-UCI_CALLID)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Progress) Options: ()
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (privacy)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (P-Preferred-Identity)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (P-Asserted-Identity)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CALLERID(all)=+49408912347686 <+49408912347686>)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPRemoveHeader) Options: (P-Asserted-Identity)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (SIPAddHeader) Options: (P-Asserted-Identity:<sip:+4940891234760@sip-trunk.telekom.de>)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(num,i)1411234311)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(num-pres,i)=allowed)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(name,i)1411234311)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Set) Options: (CONNECTEDLINE(name-pres)=allowed)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (NoOp) Options: (Dialing Line lineId73 lineName=SIP-TRUNK)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] res_agi.c: -- AGI Script Executing Application: (Dial) Options: (SIP/+4940891234760/+4971411234311,125,wWtT)
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] netsock2.c: == Using SIP RTP TOS bits 184
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] netsock2.c: == Using SIP RTP CoS mark 5
[Mar 16 09:45:18] VERBOSE[28121][C-00001f80] app_dial.c: -- Called SIP/+4940891234760/+4971411234311
[Mar 16 09:45:19] VERBOSE[14117][C-00001f80] res_rtp_asterisk.c: > 0x21ee550 -- Strict RTP learning after remote address set to: 217.123.234.52:20472
[Mar 16 09:45:19] VERBOSE[28121][C-00001f80] app_dial.c: -- SIP/+4940891234760-0000107e is ringing
[Mar 16 09:45:19] VERBOSE[28121][C-00001f80] app_dial.c: -- SIP/+4940891234760-0000107e is making progress passing it to SIP/Softphone.PT-0000107d
[Mar 16 09:45:19] VERBOSE[28121][C-00001f80] res_rtp_asterisk.c: > 0x21ee550 -- Strict RTP switching to RTP remote address 217.123.234.52:20472 as source
[Mar 16 09:45:20] VERBOSE[28121][C-00001f80] res_rtp_asterisk.c: > 0x7fd8548e44b0 -- Strict RTP learning complete - Locking on source address 192.168.77.146:4004
[Mar 16 09:45:20] VERBOSE[24821] asterisk.c: -- Remote UNIX connection
[Mar 16 09:45:20] VERBOSE[28133] asterisk.c: -- Remote UNIX connection disconnected
[Mar 16 09:45:20] VERBOSE[28121][C-00001f80] res_rtp_asterisk.c: > 0x21ee550 -- Strict RTP learning complete - Locking on source address 217.123.234.52:20472
[Mar 16 09:45:25] ERROR[14117][C-00001f80] netsock2.c: getaddrinfo("th1", "(null)", ...): Name or service not known
[Mar 16 09:45:25] WARNING[14117][C-00001f80] chan_sip.c: Invalid host name in Contact: (can't resolve in DNS) : 'th1'
[Mar 16 09:45:25] ERROR[14117][C-00001f80] netsock2.c: getaddrinfo("reg.sip-trunk.telekom.de", "(null)", ...): Name or service not known
[Mar 16 09:45:25] WARNING[14117][C-00001f80] chan_sip.c: Can't find address for host 'reg.sip-trunk.telekom.de'
[Mar 16 09:45:25] VERBOSE[28121][C-00001f80] app_dial.c: -- SIP/+4940891234760-0000107e answered SIP/Softphone.PT-0000107d
[Mar 16 09:46:31] VERBOSE[28121][C-00001f80] pbx.c: -- Executing [h@dialstart:1] NoOp("SIP/Softphone.PT-0000107d", "HC 16") in new stack
[Mar 16 09:46:31] VERBOSE[28121][C-00001f80] pbx.c: -- Executing [h@dialstart:2] Goto("SIP/Softphone.PT-0000107d", "_exit_,0") in new stack
[Mar 16 09:46:31] VERBOSE[28121][C-00001f80] pbx.c: -- Goto (dialstart,_exit_,0)
[Mar 16 09:46:31] ERROR[28121][C-00001f80] netsock2.c: getaddrinfo("reg.sip-trunk.telekom.de", "(null)", ...): Name or service not known
[Mar 16 09:46:31] WARNING[28121][C-00001f80] chan_sip.c: Can't find address for host 'reg.sip-trunk.telekom.de'
[Mar 16 09:46:31] VERBOSE[28121][C-00001f80] pbx.c: == Spawn extension (dialstart, +4971411234311, 1) exited non-zero on 'SIP/Softphone.PT-0000107d'
Alles anzeigen
support.log
[2018-03-16 09:45:18,538] [2024] ********* Call created *********
[2018-03-16 09:45:18,697] [2024] SIP/Softphone.PT-0000107d Channelstate is Up
[2018-03-16 09:45:18,713] [2024] Starting call routing : SIP/Softphone.PT-0000107d|1521189918.4262 dial number +4971411234311 CallerId click2dial: +4971411234311 <+4971411234311>
[2018-03-16 09:45:18,776] [2024] Routing call "click2dial: +4971411234311 <+4971411234311>" to number +4971411234311 over service OutgoingService
[2018-03-16 09:45:18,776] [2024] CallLeg 43fe6656-3842-4a0c-8dd2-48ea90e32f83
[2018-03-16 09:45:18,778] [2024] Found cor/lbr rules: rule1
[2018-03-16 09:45:18,779] [2024] Found lines for COR routing
[2018-03-16 09:45:18,780] [2024] - SIP/+4940891234760
[2018-03-16 09:45:18,780] [2024] - SIP/f512929147d69c1f9ad01c66eb860559feee7058
[2018-03-16 09:45:18,789] [2024] Signalnumber of caller account PojoNumber=[ accountId=0, rawnumber=0049408912347686, phoneNumberType=EXTERNAL]
[2018-03-16 09:45:18,789] [2024] SignalNumberByParticipantAndLine 0049408912347686
[2018-03-16 09:45:18,789] [2024] Signal number after calculate 0049408912347686
[2018-03-16 09:45:18,789] [2024] SipconnectDisplayNumber +49408912347686
[2018-03-16 09:45:18,790] [2024] Normalized Number +49408912347686
[2018-03-16 09:45:18,790] [2024] Signalling on Line SIP-TRUNK(1173)
[2018-03-16 09:45:18,796] [2024] CALLERID(all) +49408912347686 <+49408912347686> channel
[2018-03-16 09:45:18,801] [2024] P-Asserted-Identity <sip:+4940891234760@sip-trunk.telekom.de> sipheader
[2018-03-16 09:45:18,832] [2024] Dialing line SIP/+4940891234760 with extension +4971411234311
[2018-03-16 09:45:18,922] [2024] Dial SIP/Softphone.PT-0000107d to SIP/+4940891234760-0000107e
[2018-03-16 09:45:19,451] [2024] SIP/+4940891234760-0000107e Channelstate is Ringing
[2018-03-16 09:45:25,215] [2024] SIP/+4940891234760-0000107e Channelstate is Up
[2018-03-16 09:45:25,216] [2024] SIP/+4940891234760-0000107e Link SIP/Softphone.PT-0000107d
[2018-03-16 09:45:56,424] [2024] SIP/+4940891234760-0000107e Unlink SIP/Softphone.PT-0000107d
[2018-03-16 09:45:56,425] [2024] SIP/+4940891234760-0000107e Link SIP/Softphone.PT-0000107d
[2018-03-16 09:46:31,195] [2024] SIP/+4940891234760-0000107e Unlink SIP/Softphone.PT-0000107d
[2018-03-16 09:46:31,200] [2024] SIP/+4940891234760-0000107e Hangup Cause: Normal Clearing
[2018-03-16 09:46:31,210] [2024] Got dialstatus DialReturnCodes(hc=NORMAL_CLEARING, ds=ANSWER, cr=UNKNOWN)
[2018-03-16 09:46:31,220] [2024] SIP/Softphone.PT-0000107d Hangup Cause: Normal Clearing
[2018-03-16 09:46:31,231] [2024] ********* Call finished *********
Alles anzeigen
Eventuell hat hier ja jemand einen klugken Tipp, wie wir das Problem beseitigen können.
Danke!
Beste Grüße