MiVoice Office 400 häufig Anruf Misslungen

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TechnikOnkel

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Hallo,

ich habe hier eine "Mitel MiVoice Office 400"

Diese ist am Hauptstandort an einen LANCOM 1781A mit VDSL 100 Mbit/s angebunden.
Die Firewall des LANCOM ist deakiviert. Die Ports 5000 bis 65.000 werden von außen auf die Anlage geleitet.

Leider erhalten wir sehr häufig "Anruf Misslungen" gerade auffällig ist es, wenn der Benutzer seine Anrufe auf ein Handy weiterleitet kommt die Verbindung nicht zustande.

Wie kann ich dies weiter prüfen?

Vielen Dank.
 
Telnet auf Port 1818 und einen Rolling Trace (Predefined / Voip extended ein) mit einem Testanruf durchführen.

Da muss dann ja eine Fehlermeldung ersichtlich sein.
 
Zuletzt bearbeitet:
Wahrscheinlich hast du zu wenig Kanäle
 
Zuletzt bearbeitet von einem Moderator:
Das sieht nach Port 23 aus, nicht 1818.
 
Hallo,

wir haben eine Mitel 470 im Einsatz. Leider erhalten wir ebenfalls "Anruf Misslungen" Meldungen.
Ich hänge ein Auszug zu einer Rufnummer an. Wo liegt hier der Fehler, wo sehe ich die Kanäle vorhanden/benutzt?

SIP 14:58:59.572: TRANSPORT : [0xbb164e8]IaSipTcpSocketC::handleSocketData:10.20.0.102:60682:TCP
SIP 14:58:59.572: TRANSPORT : IaSipTcpSocketC::startInactivityTimer
SIP 14:58:59.572: TRANSPORT : Started inactivity timer
SIP 14:58:59.572: TRANSPORT : IaSipTransportC::handleData:10.20.0.102:60682:TCP
SIP 14:58:59.572: TRANSCEIVER : isFreshRequest: No transaction found
SIP 14:58:59.573: TRANSCEIVER : LAN > UA: "INVITE" request [1986301408 INVITE] received from [10.20.0.102:60682:TCP]
SIP 14:58:59.573: TRANSCEIVER : INVITE sip:[email protected]:5060;user=phone SIP/2.0
SIP 14:58:59.573: TRANSCEIVER : Via: SIP/2.0/TCP 10.20.0.102;branch=z9hG4bK14a829e5fc803e4d6
SIP 14:58:59.573: TRANSCEIVER : Max-Forwards: 70
SIP 14:58:59.573: TRANSCEIVER : From: "Nachname, Vorname" <sip:[email protected]:5060>;tag=462b432273
SIP 14:58:59.573: TRANSCEIVER : To: <sip:[email protected]:5060;user=phone>
SIP 14:58:59.573: TRANSCEIVER : Call-ID: ce4816991c3bdf39
SIP 14:58:59.573: TRANSCEIVER : CSeq: 1986301408 INVITE
SIP 14:58:59.573: TRANSCEIVER : Accept-Language: de
SIP 14:58:59.573: TRANSCEIVER : Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
SIP 14:58:59.573: TRANSCEIVER : Allow-Events: aastra-xml, vdp-session, talk, hold, conference, LocalModeStatus
SIP 14:58:59.573: TRANSCEIVER : Contact: "Nachname, Vorname" <sip:[email protected]:5060;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-08000FCEDB69>"
SIP 14:58:59.573: TRANSCEIVER : Session-Expires: 3600
SIP 14:58:59.573: TRANSCEIVER : Supported: path, gruu, 100rel, replaces, timer
SIP 14:58:59.573: TRANSCEIVER : User-Agent: Mitel 6930/5.1.0.2047
SIP 14:58:59.573: TRANSCEIVER : Content-Type: application/sdp
SIP 14:58:59.573: TRANSCEIVER : Content-Length: 410
SIP 14:58:59.573: TRANSCEIVER :
SIP 14:58:59.573: TRANSCEIVER : v=0
SIP 14:58:59.573: TRANSCEIVER : o=MxSIP 0 1 IN IP4 10.20.0.102
SIP 14:58:59.573: TRANSCEIVER : s=SIP Call
SIP 14:58:59.573: TRANSCEIVER : c=IN IP4 10.20.0.102
SIP 14:58:59.574: TRANSCEIVER : t=0 0
SIP 14:58:59.574: TRANSCEIVER : m=audio 3000 RTP/AVP 8 9 0 18 101
SIP 14:58:59.574: TRANSCEIVER : a=rtpmap:8 PCMA/8000
SIP 14:58:59.574: TRANSCEIVER : a=rtpmap:9 G722/8000
SIP 14:58:59.574: TRANSCEIVER : a=rtpmap:0 PCMU/8000
SIP 14:58:59.574: TRANSCEIVER : a=rtpmap:18 G729/8000
SIP 14:58:59.574: TRANSCEIVER : a=rtpmap:101 telephone-event/8000
SIP 14:58:59.574: TRANSCEIVER : a=silenceSupp:eek:ff - - - -
SIP 14:58:59.574: TRANSCEIVER : a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vnx1ViFTPXZlOlxvdXU1aUM/MVk8MVBeJDB6fUQx
SIP 14:58:59.574: TRANSCEIVER : a=fmtp:18 annexb=no
SIP 14:58:59.574: TRANSCEIVER : a=fmtp:101 0-15
SIP 14:58:59.574: TRANSCEIVER : a=ptime:20
SIP 14:58:59.574: TRANSCEIVER : a=sendrecv
SIP 14:58:59.575: TRANSCEIVER : transmitSipMessage(10.20.0.102:60682:TCP)
SIP 14:58:59.575: TRANSCEIVER : Sending Outgoing SIP message through the ALG
SIP 14:58:59.575: SIP_ALG : IaSipAlgC::handleOutgoingMessage(100 Trying,10.20.0.102:60682:TCP)
SIP 14:58:59.575: SIP_ALG : IaSipAlgC::updateOutgoingMessage(10.20.0.102:60682:TCP)
SIP 14:58:59.576: SIP_ALG : Found existing entity:
SIP 14:58:59.576: SIP_ALG : Original Src IP : 10.20.0.102:60682:TCP
SIP 14:58:59.576: SIP_ALG : Original Contact IP: 10.20.0.102:5060:TCP
SIP 14:58:59.576: SIP_ALG : New Contact IP : 10.20.0.102:5060:TCP
SIP 14:58:59.576: SIP_ALG : Original Media IP : 10.20.0.102:3000:UDPorTCP
SIP 14:58:59.576: SIP_ALG : New Media IP : 10.20.0.102:3000:UDPorTCP
SIP 14:58:59.576: SIP_ALG : RFC3581 IP : 0.0.0.0:0:UDPorTCP
SIP 14:58:59.576: SIP_ALG : Behind NAT : no
SIP 14:58:59.576: SIP_ALG : External Subscriber: no
SIP 14:58:59.576: SIP_ALG : Using local PBX address:10.10.0.10:5060:TCP
SIP 14:58:59.576: TRANSCEIVER : transmitSipMessage() changed the transport to 'TCP'
SIP 14:58:59.577: TRANSCEIVER : LAN < UA: "100 Trying" status [1986301408 INVITE] sent to [10.20.0.102:60682:TCP]
SIP 14:58:59.577: TRANSCEIVER : SIP/2.0 100 Trying
SIP 14:58:59.577: TRANSCEIVER : Via: SIP/2.0/TCP 10.20.0.102;branch=z9hG4bK14a829e5fc803e4d6
SIP 14:58:59.577: TRANSCEIVER : To: <sip:[email protected]:5060;user=phone>
SIP 14:58:59.577: TRANSCEIVER : From: "Nachname, Vorname" <sip:[email protected]:5060>;tag=462b432273
SIP 14:58:59.577: TRANSCEIVER : Call-ID: ce4816991c3bdf39
SIP 14:58:59.577: TRANSCEIVER : CSeq: 1986301408 INVITE
SIP 14:58:59.577: TRANSCEIVER : Content-Length: 0
SIP 14:58:59.577: TRANSCEIVER :
SIP 14:58:59.577: TRANSCEIVER : IaSipTransceiverC::transmitSipMessage() encoding the message
SIP 14:58:59.577: TRANSPORT : IaSipTransportC::send:10.20.0.102:60682:TCP
SIP 14:58:59.577: TRANSPORT : IaSipTransportC::sendOverTcp
SIP 14:58:59.577: TRANSPORT : Found open TCP socket: [0xbb164e8]
SIP 14:58:59.577: TRANSPORT : [0xbb164e8]IaSipTcpSocketC::sendData(10.20.0.102:60682:TCP)
SIP 14:58:59.577: TRANSPORT : IaSipTcpSocketC::sendPendingData
SIP 14:58:59.577: TRANSPORT : [0xbb164e8]IaSipSocketC::sendData:10.20.0.102:60682:TCP
SIP 14:58:59.578: TRANSPORT : Data sent (ok to send more)
SIP 14:58:59.578: TRANSPORT : IaSipTcpSocketC::startInactivityTimer
SIP 14:58:59.578: TRANSPORT : Started inactivity timer
SIP 14:58:59.578: TRANSCEIVER : Incoming SIP message was not sent through the ALG
SIP 14:58:59.579: UA_SIP_MSG : LAN > UA: "INVITE" request [1986301408 INVITE] received from [10.20.0.102:60682:TCP]
SIP 14:58:59.579: UA_SIP_MSG : INVITE sip:[email protected]:5060;user=phone SIP/2.0
SIP 14:58:59.579: UA_SIP_MSG : Via: SIP/2.0/TCP 10.20.0.102;branch=z9hG4bK14a829e5fc803e4d6
SIP 14:58:59.579: UA_SIP_MSG : Max-Forwards: 70
SIP 14:58:59.579: UA_SIP_MSG : From: "Nachname, Vorname" <sip:[email protected]:5060>;tag=462b432273
SIP 14:58:59.579: UA_SIP_MSG : To: <sip:[email protected]:5060;user=phone>
SIP 14:58:59.579: UA_SIP_MSG : Call-ID: ce4816991c3bdf39
SIP 14:58:59.579: UA_SIP_MSG : CSeq: 1986301408 INVITE
SIP 14:58:59.579: UA_SIP_MSG : Accept-Language: de
SIP 14:58:59.579: UA_SIP_MSG : Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
SIP 14:58:59.579: UA_SIP_MSG : Allow-Events: aastra-xml, vdp-session, talk, hold, conference, LocalModeStatus
SIP 14:58:59.579: UA_SIP_MSG : Contact: "Nachname, Vorname" <sip:[email protected]:5060;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-08000FCEDB69>"
SIP 14:58:59.579: UA_SIP_MSG : Session-Expires: 3600
SIP 14:58:59.579: UA_SIP_MSG : Supported: path, gruu, 100rel, replaces, timer
SIP 14:58:59.579: UA_SIP_MSG : User-Agent: Mitel 6930/5.1.0.2047
SIP 14:58:59.579: UA_SIP_MSG : Content-Type: application/sdp
SIP 14:58:59.579: UA_SIP_MSG : Content-Length: 410
SIP 14:58:59.579: UA_SIP_MSG :
SIP 14:58:59.579: UA_SIP_MSG : v=0
SIP 14:58:59.580: UA_SIP_MSG : o=MxSIP 0 1 IN IP4 10.20.0.102
SIP 14:58:59.580: UA_SIP_MSG : s=SIP Call
SIP 14:58:59.580: UA_SIP_MSG : c=IN IP4 10.20.0.102
SIP 14:58:59.580: UA_SIP_MSG : t=0 0
SIP 14:58:59.580: UA_SIP_MSG : m=audio 3000 RTP/AVP 8 9 0 18 101
SIP 14:58:59.580: UA_SIP_MSG : a=rtpmap:8 PCMA/8000
SIP 14:58:59.580: UA_SIP_MSG : a=rtpmap:9 G722/8000
SIP 14:58:59.580: UA_SIP_MSG : a=rtpmap:0 PCMU/8000
SIP 14:58:59.580: UA_SIP_MSG : a=rtpmap:18 G729/8000
SIP 14:58:59.580: UA_SIP_MSG : a=rtpmap:101 telephone-event/8000
SIP 14:58:59.580: UA_SIP_MSG : a=silenceSupp:eek:ff - - - -
SIP 14:58:59.580: UA_SIP_MSG : a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vnx1ViFTPXZlOlxvdXU1aUM/MVk8MVBeJDB6fUQx
SIP 14:58:59.580: UA_SIP_MSG : a=fmtp:18 annexb=no
SIP 14:58:59.580: UA_SIP_MSG : a=fmtp:101 0-15
SIP 14:58:59.580: UA_SIP_MSG : a=ptime:20
SIP 14:58:59.580: UA_SIP_MSG : a=sendrecv
SIP 14:58:59.581: UA_SESSION : Invite from None will be checked.
IBT 14:58:59.582: BCS: Client: iaSipProc Bcs: 411 MT: ReserveNew
SIP 14:58:59.583: UA_SESSION : 0bebf5e8: Media Session 411 has been created
SIP 14:58:59.583: USER_AGENT : Added Media session(411) [callId="ce4816991c3bdf39", localTag="AI22E6BCB17F5DFE79", remoteTag="462b432273"] to SessionIdMap sessP = 0bebf5e8
SIP 14:58:59.583: USER_AGENT : Added Media session(411) [callId="ce4816991c3bdf39", localTag="AI22E6BCB17F5DFE79", remoteTag="462b432273"] to DialogIdMap sessP = 0bebf5e8
SIP 14:58:59.583: USER_AGENT : Media Session 0bebf5e8 found for incoming REQUEST where CSeq="1986301408 INVITE"
SIP 14:58:59.583: UA_SESSION : 0bebf5e8: Remote peer allows "UPDATE"
SIP 14:58:59.583: UA_SESSION : IaSipUaBaseSessionC::checkSupported.
SIP 14:58:59.584: UA_SESSION : IaSipUaMediaSessionC::checkSupported() SR ENabled in monitor
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: Remote peer supports "replaces"
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: Session Replacement enabled
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: Remote peer supports "timer"
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: getNextHop(), myPriRemoteAddr: 10.20.0.102:60682:TCP
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: getNextHop(), returning nextHopAddr: 10.20.0.102:60682:TCP
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: Incoming message has updated the stored contact URI to [sip:[email protected]:5060;transport=tcp]
SIP 14:58:59.584: UA_SESSION : 0bebf5e8: getNextHop(), myPriRemoteAddr: 10.20.0.102:60682:TCP
SIP 14:58:59.585: UA_SESSION : 0bebf5e8: getNextHop(): taken the URL from myContactUri: [sip:[email protected]:5060;transport=tcp]
SIP 14:58:59.585: UA_SESSION : 0bebf5e8: getNextHop(): host to resolve: [10.20.0.102]
SIP 14:58:59.585: UA_SESSION : getNextHop(), setting port to: 5060
SIP 14:58:59.585: UA_SESSION : 0bebf5e8: getNextHop(), returning nextHopAddr: 10.20.0.102:5060:TCP
SIP 14:58:59.585: UA_SESSION : 0bebf5e8: Incoming message has updated the routing, next hop is now [10.20.0.102:5060:TCP]
SIP 14:58:59.585: UA_SESSION : sessionTimerHandleRequest() RValue: 3600 RFresh Entity yes
SIP 14:58:59.585: UA_SESSION : 0bebf5e8: Set myLastHandledReqSeqNum to 1986301408
SIP 14:58:59.585: UA_MEDIASESS: 0bebf5e8: handleSipInvite(Bcs: 411) in state: Idle
SIP 14:58:59.585: UA_MEDIASESS: Call not from provider, set displayName!!!
SIP 14:58:59.586: UA_MEDIASESS: 0bebf5e8: In getMediaInfoFromSDP(Bcs: 411), sdpOffer=true
SIP 14:58:59.586: UA_MEDIASESS: - MediaSectionList(Bcs: 411): Audio: [-1], Video: [-1], TCPImage: [-1], UDPTLImage: [-1], Unknown: [-1]
SIP 14:58:59.586: UA_MEDIASESS: Non-secure signaling: SRTP suppressed.
SIP 14:58:59.586: UA_MEDIASESS: 0bebf5e8: Return from getMediaInfoFromSDP(Bcs: 411): remoteMediaAddr=[Offer: 10.20.0.102;voip[1]=3000(sendrecv)], sdpCheckResult: SupportedSdp
SIP 14:58:59.586: UA_MEDIASESS: - MediaSectionList(Bcs: 411): Audio: [1], Video: [-1], TCPImage: [-1], UDPTLImage: [-1], Unknown: [-1]
SIP 14:58:59.586: UA_MEDIASESS: 0bebf5e8: processRemoteMedia(): ourSrtpMode: Enabled
SIP 14:58:59.586: UA_MEDIASESS: 0bebf5e8: processRemoteMedia(): remote media changed!
SIP 14:58:59.587: USER_AGENT : Call not from provider
SIP 14:58:59.587: USER_AGENT : Call not from remote node
SIP 14:58:59.587: USER_AGENT : Terminal wants to handle diversions
SIP 14:58:59.587: UA_MEDIASESS: setRemotePtyInfo(), currently: myRemoteNr: [], myRemoteName: []
SIP 14:58:59.587: UA_MEDIASESS: Changed, myRemoteNr: [58-93], myRemoteName: [Nachname, Vorname]
SIP 14:58:59.587: UA_MEDIASESS: Called Party: [00821207124578] taken from: Request-Line
SIP 14:58:59.588: UA_MEDIASESS: 0bebf5e8: Media Session Bcs: 411 changed state from 'Idle' to 'Connecting'
SIP 14:58:59.588: UA-ER_IFACE : UA => ER: establishCall(Bcs: 411), cdPty=[00821207124578], cgPty=[58-93], cgPtyName=[Nachname, Vorname], clir=false, sipEpId=None, termId=105
SIP 14:58:59.588: UA-ER_IFACE : mediaInfo=[Offer: 10.20.0.102;voip[1]=3000(sendrecv)]
IBT 14:58:59.671: BCS: Client: erPlsProc Bcs: 411 MT: Reserve
SIP 14:58:59.671: ER_S_CALLHDL: Establish incoming call (SIP)=> Bcs: 411 - did NOT remove callingPtyName: Nachname, Vorname, cgPtyNb: 58-93!
SIP 14:58:59.671: ER_S_CALLHDL: Call Handler has created session => Bcs: 411, sessionP=C078E58, number of sessions is now 9
SIP 14:58:59.671: ER_S_CALLHDL: Establish incoming call (SIP)=> Bcs: 411
SIP 14:58:59.672: ER-PBX_IFACE: ER > MR : MediaOffer(Bcs: 411) in state: Idle, mediaState: Idle, media=[Offer: 10.20.0.102;voip[1]=3000(sendrecv)]
MR 14:58:59.672: _SIG_: ________________________________________
MR 14:58:59.673: _SIG_:
MR 14:58:59.673: _SIG_: ==>> MediaOffer (from erPlsProc) {
MR 14:58:59.673: _SIG_: Bcs: 411; p/ch: 12/105
MR 14:58:59.673: _SIG_: MediaInfo :
MR 14:58:59.673: _SIG_: Offer:
MR 14:58:59.673: _SIG_: mediaAddr : 10.20.0.102
MR 14:58:59.673: _SIG_: ===== Voice =====
MR 14:58:59.673: _SIG_: SDP Position : 1
MR 14:58:59.673: _SIG_: Media port : 3000
MR 14:58:59.673: _SIG_: Direction : sendrecv
MR 14:58:59.673: _SIG_: CodecList : {PCMA(8)<<G722(9)<<PCMU(0)<<G729(18);20ms}
MR 14:58:59.673: _SIG_: AuxMediaAttr : Tel. events: 101, CN: N, VAD: N, RTCP: None
MR 14:58:59.673: _SIG_: }
SIP 14:58:59.675: ER-PBX_IFACE: Session (Bcs: 411) changed mediaState from: Idle to: OfferFromUser
SIP 14:58:59.675: ER_S_SESSION: Establish incoming call => Bcs: 411, calledPty: [00821207124578]
SIP 14:58:59.676: ER-PBX_IFACE: ER > PBX: Setup(Bcs: 411), pref=12, msn=105, ch=0, lineId=0
SIP 14:58:59.676: ER_S_SESSION: IncomingSession(Bcs: 411) in state: Idle, checkSetFacilitiesInSetup()
SIP 14:58:59.676: ER_S_SESSION: Call Bcs: 411: Changed state from Idle to Initiate
IBT 14:58:59.676: VP: << Port:SIP-Int Cr:B019B Bcs: 411 MT: Setup SUT:098|000|000
IBT BC > 04 03 90 90 a3
IBT CHAN ID > 18 01 89
IBT exclusive B1
IBT SEND COMPL > a1
IBT PROG IND > 1e 02 80 83
IBT [user] origination is non-ISDN
IBT SHIFT 7 NL > 9f
IBT FAC INFO > 59 3b 00 0b 01 00 00 00 ff ff ff 45 52 5f 53 5f 53 45 53
IBT 53 49 4f 4e 3a 20 43 61 6c 00 20 42 63 73 3a 25 34 64 3a
IBT 20 43 68 61 6e 67 65 64 20 73 74 61 74 65 20 66 72 6f 6d
IBT 20 25 73 20
IBT SIP Div.Header: CallerDivHandling=Yes, DivInfoId=0, RedPtyName=, RedPtyNum=
IBT CD PTY NMB > 70 0e 80 30 30 38 32 31 32 30 37 38 38 34 32 38
IBT Unknown/Unknown [00821207124578]
IBT 14:58:59.677: XA: ========================================================
IBT 14:58:59.677: XA: << SetupInd Bcs: 411 pref: 12
IBT 14:58:59.677: BCS: Client: xaNcpProc Bcs: 411 MT: Reserve
IBT 14:58:59.678: XA: SINGLELINE SIP
IBT 14:58:59.678: XA: no existing node found for port=12 chan=105
IBT 14:58:59.678: XA: SINGLELINE SIP / Exclusive
IBT 14:58:59.678: XA: xaNcxAllocChFor(), Bcs: 411, chan: 105, representLine: 0
IBT 14:58:59.679: CON: Con[00000000] allocated Bcs: 411 BCh=105 pref=012
IBT 14:58:59.680: XA: xaNcaExamine1Selection : selection is 00821207124578
IBT 14:58:59.680: XA: The number is of type: Global Exchange Access code
IBT 14:58:59.680: XA: external Access size is: 1
IBT 14:58:59.680: XA: XaNcaExaAnalExtGlobal
IBT 14:58:59.681: XA: HuntLine: Bcs: 411 permission-check=1 perm-sref=58/(93)
IBT 14:58:59.681: XA: search line to call 00821207124578
IBT 14:58:59.681: XA: permissionSet in node is 1
IBT 14:58:59.681: XA: ttl: route:2 pref:0 chGr:0 hc:0
IBT 14:58:59.682: XA: decRoute:FirstLine: bundle 2 found. start looking for line with pref 0
IBT 14:58:59.682: XA: decLine: bundle:2 pref:0 chGr:255
IBT 14:58:59.682: XA: init FirstHunted, lastAssign or FirstHunted are not in TrGr
IBT 14:58:59.682: XA: decLine: look for prev ext port. start-pref:28 found-pref:27
IBT 14:58:59.682: XA: primary found, pref:27 chGr:0
IBT 14:58:59.683: XA: decLine: return pref:27 channelGroup: 0
IBT 14:58:59.683: XA: test line: route=2 bundle=2 pref=27 channelGroup=1
IBT 14:58:59.684: CON: >> Reserve: srcBcs: 411 shrBcs:65535 srcPrf:12 srcMsn:105 desPrf:27 desMsn:30 desNum:4982120788428
MR 14:58:59.684: _SIG_: ________________________________________
MR 14:58:59.684: _SIG_:
MR 14:58:59.684: _SIG_: ==>> ReserveRouteReq {
MR 14:58:59.684: _SIG_: source : Bcs: 411; p/ch: 12/105
MR 14:58:59.684: _SIG_: destination : Bcs:65535; p/ch: 1/30
MR 14:58:59.684: _SIG_: reuseRouteId : 0
MR 14:58:59.684: _SIG_: reservationType : VoIP
MR 14:58:59.684: _SIG_: }
MR 14:58:59.685: _SIG_: ________________________________________
MR 14:58:59.685: _SIG_:
MR 14:58:59.685: _SIG_: <<== ReserveRouteResp {
MR 14:58:59.686: _SIG_: routeId : 669
MR 14:58:59.686: _SIG_: respCode : Success
MR 14:58:59.686: _SIG_: }
MR 14:58:59.686: _SIG_: ________________________________________
MR 14:58:59.686: _SIG_:
MR 14:58:59.686: _SIG_: ==>> ReserveRouteReq {
MR 14:58:59.686: _SIG_: source : Bcs:65535; p/ch: 1/30
MR 14:58:59.686: _SIG_: destination : Bcs:65535; p/ch: 27/30
MR 14:58:59.686: _SIG_: reuseRouteId : 0
MR 14:58:59.686: _SIG_: reservationType : VoIP
MR 14:58:59.686: _SIG_: }
MR 14:58:59.687: _SIG_: ________________________________________
MR 14:58:59.687: _SIG_:
MR 14:58:59.687: _SIG_: <<== ReserveRouteResp {
MR 14:58:59.687: _SIG_: routeId : 670
MR 14:58:59.687: _SIG_: respCode : Success
MR 14:58:59.687: _SIG_: }
IBT 14:58:59.687: CON: >> Unreserve: Bcs: 411 desPrf:27 desNum:4982120788428 destMsn:30
MR 14:58:59.688: _SIG_: ________________________________________
MR 14:58:59.688: _SIG_:
MR 14:58:59.688: _SIG_: ==>> UnreserveRouteReq {
MR 14:58:59.688: _SIG_: routeId : 669
MR 14:58:59.688: _SIG_: }
MR 14:58:59.688: _SIG_: #### FreeSwitchObject not sent, NOT used by switching
MR 14:58:59.688: _SIG_: ________________________________________
MR 14:58:59.688: _SIG_:
MR 14:58:59.688: _SIG_: <<== MrMonVoIPLoadInd (1 x) {
MR 14:58:59.688: _SIG_: nodeId : 0
MR 14:58:59.688: _SIG_: nrOfUsedVoIPCh : 9
MR 14:58:59.688: _SIG_: nrOfVoIPCh : 32
MR 14:58:59.688: _SIG_: peakOfUsedVoIPCh : 31
MR 14:58:59.688: _SIG_: }
MR 14:58:59.689: _SIG_: ________________________________________
MR 14:58:59.689: _SIG_:
MR 14:58:59.689: _SIG_: ==>> UnreserveRouteReq {
MR 14:58:59.689: _SIG_: routeId : 670
MR 14:58:59.689: _SIG_: }
MR 14:58:59.689: _SIG_: #### FreeSwitchObject not sent, NOT used by switching
MR 14:58:59.689: _SIG_: ________________________________________
MR 14:58:59.689: _SIG_:
MR 14:58:59.689: _SIG_: <<== MrMonVoIPLoadInd (1 x) {
MR 14:58:59.689: _SIG_: nodeId : 0
MR 14:58:59.689: _SIG_: nrOfUsedVoIPCh : 8
MR 14:58:59.689: _SIG_: nrOfVoIPCh : 32
MR 14:58:59.689: _SIG_: peakOfUsedVoIPCh : 31
MR 14:58:59.689: _SIG_: }
MR 14:58:59.689: _SIG_: ##### Removed unused SIP node: 6330
IBT 14:58:59.690: XA: go for this line on port=27
IBT 14:58:59.690: XA: successful: 1
IBT 14:58:59.690: XA: xaNcxAllocChFor(), Bcs: 411, chan: 105, representLine: 0
IBT 14:58:59.692: XA: The number queried from the name server is:[00821207124578] and name to be updated:[ ]
IBT 14:58:59.693: XA-TP: ListControlInfo: UserId:93 Type=Redial Command=Insert Reason=Undef partner:00821207124578
IBT 14:58:59.693: BCS: Client: xaNcpProc Bcs: 412 MT: ReserveNew
IBT 14:58:59.694: CON: >> Reserve: srcBcs: 411 shrBcs:65535 srcPrf:12 srcMsn:105 desPrf:27 desMsn:30 desNum:4982120788428
MR 14:58:59.694: _SIG_: ________________________________________
MR 14:58:59.694: _SIG_:
MR 14:58:59.694: _SIG_: ==>> ReserveRouteReq {
MR 14:58:59.694: _SIG_: source : Bcs: 411; p/ch: 12/105
MR 14:58:59.694: _SIG_: destination : Bcs:65535; p/ch: 1/30
MR 14:58:59.695: _SIG_: reuseRouteId : 0
MR 14:58:59.695: _SIG_: reservationType : VoIP
MR 14:58:59.695: _SIG_: }
MR 14:58:59.696: _SIG_: ________________________________________
MR 14:58:59.696: _SIG_:
MR 14:58:59.696: _SIG_: <<== ReserveRouteResp {
MR 14:58:59.696: _SIG_: routeId : 671
MR 14:58:59.696: _SIG_: respCode : Success
MR 14:58:59.696: _SIG_: }
MR 14:58:59.696: _SIG_: ________________________________________
MR 14:58:59.696: _SIG_:
MR 14:58:59.696: _SIG_: ==>> ReserveRouteReq {
MR 14:58:59.696: _SIG_: source : Bcs:65535; p/ch: 1/30
MR 14:58:59.696: _SIG_: destination : Bcs:65535; p/ch: 27/30
MR 14:58:59.696: _SIG_: reuseRouteId : 0
MR 14:58:59.696: _SIG_: reservationType : VoIP
MR 14:58:59.696: _SIG_: }
MR 14:58:59.697: _SIG_: ________________________________________
MR 14:58:59.697: _SIG_:
MR 14:58:59.697: _SIG_: <<== ReserveRouteResp {
MR 14:58:59.697: _SIG_: routeId : 672
MR 14:58:59.697: _SIG_: respCode : Success
MR 14:58:59.697: _SIG_: }
IBT 14:58:59.698: CON: Con[00000000] allocated Bcs: 412 BCh=30 pref=027
IBT 14:58:59.698: CON: >> Update Destination: srcBcs: 411 destBcs: 412
MR 14:58:59.698: _SIG_: ________________________________________
MR 14:58:59.699: _SIG_:
MR 14:58:59.699: _SIG_: ==>> UpdateRouteReq {
MR 14:58:59.699: _SIG_: routeId : 671
MR 14:58:59.699: _SIG_: source : Bcs: 411
MR 14:58:59.699: _SIG_: destination: Bcs: 412
MR 14:58:59.699: _SIG_: }
MR 14:58:59.699: _SIG_: ________________________________________
MR 14:58:59.699: _SIG_:
MR 14:58:59.699: _SIG_: ==>> UpdateRouteReq {
MR 14:58:59.699: _SIG_: routeId : 672
MR 14:58:59.699: _SIG_: source : Bcs: 411
MR 14:58:59.699: _SIG_: destination: Bcs: 412
MR 14:58:59.699: _SIG_: }
IBT 14:58:59.699: XA: selectionComplete bit of if-pointer is set
IBT 14:58:59.700: XA: xaRouSmartDdiOutgoingLookUp() xaroutelerouting.cc ddiPlanId:2 internalNbrStr:58
IBT 14:58:59.700: XA: resultingDdiNumber:004991153992358 found in case of ddi cache lookup
IBT 14:58:59.700: XA: - - - - - - - - - - - - - - - - - - - - - - - - - - - -
IBT 14:58:59.700: XA: >> SetupReq Bcs: 412 pref: 27 ns:17
IBT 14:58:59.701: VP: >> Port:SIP-Ext Cr:B019C Cid:52802 Bcs: 412 MT: Setup
IBT BC > 04 03 90 90 a3
IBT CHAN ID > 18 03 a9 83 9f
IBT exclusive B30
IBT PROG IND > 1e 02 80 83
IBT [user] origination is non-ISDN
IBT CG PTY NMB > 6c 11 00 80 30 30 34 39 39 31 31 35 33 39 39 32 33 35 38
IBT Unknown/Unknown [004991153992358]
IBT CG PTY NMB > 6c 11 c9 42 61 64 68 6f 72 6e 2c 20 4d 69 63 68 61 65 6c
IBT Private/Level0 [Nachname, Vorname]
IBT CD PTY NMB > 70 0d 80 30 38 32 31 32 30 37 38 38 34 32 38
IBT Unknown/Unknown [0821207124578]
IBT SEND COMPL > a1
IBT 14:58:59.701: XA: - - - - - - - - - - - - - - - - - - - - - - - - - - - -
IBT 14:58:59.701: XA: >> CallProcReq Bcs: 411 pref: 12 ns: 9
IBT 14:58:59.701: VP: >> Port:SIP-Int Cr:B019B Cid:52802 Bcs: 411 MT: Call Proc SUT:098|093|105
IBT CHAN ID > 18 01 8b
IBT exclusive any channel
IBT CD PTY NMB > 70 03 ca 0a 20
IBT Aastra400/Name [ ]
IBT CD PTY NMB > 70 12 40 00 07 bb 30 2d 30 38 32 31 32 30 37 38 38 34 32
IBT 38
IBT Aastra400/Number [0-0821207124578] PartyId=955
SWI 2022:05:11 14:58:59.679.119 Start NodeId 00 from Proc: rmAllMainProc
SWI [SwiMgr] Create Switch Obj, pRef,chan:012,104, Type SipSubs, Number 58
SWI [Acs ] Create, SO 41428
SWI [Ace ] Create, SO 41428, PortRef 12, BCh 104
SWI [Acs ] Link 1, SO 41428
SWI 2022:05:11 14:58:59.679.428 duration in us: 00000309 End NodeId 00
SWI 2022:05:11 14:58:59.698.150 Start NodeId 00 from Proc: rmAllMainProc
SWI [SwiMgr] Create Switch Obj, pRef,chan:027,29, Type SipTrunk, Number unknwon
SWI [Acs ] Create, SO 41429
SWI [Ace ] Create, SO 41429, PortRef 27, BCh 29
SIP 14:58:59.719: ER_S_CALLHDL: ER < PBX: establishCall(Bcs: 412)
SIP 14:58:59.719: ER-PBX_IFACE: ER < PBX: EstablishOutgoingCall(Bcs: 412) received
IBT 14:58:59.719: BCS: Client: erPlsProc Bcs: 412 MT: Reserve
SIP 14:58:59.720: ER_S_CALLHDL: Call Handler has created session => Bcs: 412, sessionP=BF84D38, number of sessions is now 10
SWI [Acs ] Link 1, SO 41429
SWI 2022:05:11 14:58:59.698.415 duration in us: 00000265 End NodeId 00
SIP 14:58:59.722: ER_S_CALLHDL: Establish outgoing call (PBX)=> Bcs: 412, Port:27, Channel:30, termId: 0
SIP 14:58:59.722: ER_S_SESSION: ErPlsOutgoingSessionC::establishOutgoing: termId=0
SIP 14:58:59.722: ER-PBX_IFACE: ER > PBX: SetupAck(Bcs: 412), pref=27, chanId=30
SIP 14:58:59.723: ER_S_SESSION: Establish outgoing call => Bcs: 412 send to external - CdPtyNb is complete!
SIP 14:58:59.723: ER-PBX_IFACE: ER > MR : MediaRequest(Bcs: 412) in state: Idle, mediaState: Idle
MR 14:58:59.723: _SIG_: ________________________________________
MR 14:58:59.723: _SIG_:
MR 14:58:59.723: _SIG_: ==>> GetPeerIPDetailsReq (from erPlsProc) {
MR 14:58:59.723: _SIG_: Bcs: 412; p/ch: 27/30
MR 14:58:59.723: _SIG_: }
MR 14:58:59.724: _SIG_: ________________________________________
MR 14:58:59.724: _SIG_:
MR 14:58:59.724: _SIG_: <<== GetPeerMediaResp (MediaOffer to erPlsProc) {
MR 14:58:59.724: _SIG_: Bcs: 412; p/ch: 27/30
MR 14:58:59.724: _SIG_: MediaInfo :
MR 14:58:59.724: _SIG_: Offer:
MR 14:58:59.724: _SIG_: mediaAddr : 10.10.0.11
MR 14:58:59.724: _SIG_: ===== Voice =====
MR 14:58:59.724: _SIG_: SDP Position : 1
MR 14:58:59.724: _SIG_: Media port : 5022
MR 14:58:59.724: _SIG_: Direction : sendrecv
MR 14:58:59.724: _SIG_: CodecList : {PCMA(8)<<PCMU(0)<<CN(13);20ms}
MR 14:58:59.724: _SIG_: AuxMediaAttr : Tel. events: 101, CN: N, VAD: N, RTCP: 5023 XR: 10.10.0.11
MR 14:58:59.724: _SIG_: }
SIP 14:58:59.724: ER-PBX_IFACE: ER < MR : MediaResponse (MediaRespOffer), media=[Offer: 10.10.0.11;voip[1]=5022(sendrecv)]
SIP 14:58:59.724: ER-PBX_IFACE: Session (Bcs: 412) changed mediaState from: Idle to: OfferFromPbx
SIP 14:58:59.725: ER-PBX_IFACE: Session (Bcs: 412) changed mediaState from: OfferFromPbx to: OfferToUser
SIP 14:58:59.725: UA-ER_IFACE : ER => UA: establishOutgoingCall(Bcs: 412)
media_addr=[Offer: 10.10.0.11;voip[1]=5022(sendrecv)]
calledPtyNb=[0821207124578], callingPtyNb=[004991153992358], callingPtyName=[Nachname, Vorname]
wishedClip=[],
CLIR=false, termId=0, sipEpId=1(SipProvider), alertPatt=0, annMode=4
transitCounter=255, emergencyCallId=Bcs:65535
SIP 14:58:59.725: UA-ER_IFACE : No Diversion Information
SIP 14:58:59.725: USER_AGENT : handleEstablishOutgoingCall(Media Bcs: 412)
SIP 14:58:59.725: USER_AGENT : sipEpId: [1(SipProvider)], remoteNodeId: 0, providerId: 1
SIP 14:58:59.725: UA_MEDIASESS: handleEstablishOutgoingCall(Bcs: 412) : Endpoint 65537 has pref codec: Undef and Codec Mode=Off, codec List={PCMA(8)<<PCMU(0)<<CN(13);20ms}
SIP 14:58:59.725: USER_AGENT : No diversions
SIP 14:58:59.725: USER_AGENT : Outgoing external SIP call, providerId=1
SIP 14:58:59.726: USER_AGENT : Using configured realmName [sip-trunk.telekom.de] as destination domain.
SIP 14:58:59.726: USER_AGENT : Using configured realmName [sip-trunk.telekom.de] as destination domain.
SIP 14:58:59.726: USER_AGENT : Looking for account to use for the outgoing call
SIP 14:58:59.726: USER_AGENT : Looking for an account for the calling party
SIP 14:58:59.726: USER_AGENT : User doesn't have a dedicated account, checking for a default account
SIP 14:58:59.726: USER_AGENT : Found default account
SIP 14:58:59.726: USER_AGENT : Updating 'From'-field
SIP 14:58:59.726: USER_AGENT : Updating 'PPI'-header with SystemClip
SIP 14:58:59.726: USER_AGENT : Using configured realmName [sip-trunk.telekom.de] as destination domain.
SIP 14:58:59.726: USER_AGENT : Before calling establishOutgoingCall, From ="Gudeco Nuernberg" <sip:[email protected]>, To=sip:[email protected], destUrl=0821207124578, preferredIdentity=+4991153992358
SIP 14:58:59.727: UA_MEDIASESS: IaSipUserAgentC::establishOutgoingCall(Bcs: 412), from: ["Gudeco Nuernberg" <sip:[email protected]>], to: [sip:[email protected]]
IBT 14:58:59.727: BCS: Client: iaSipProc Bcs: 412 MT: Reserve
SIP 14:58:59.727: UA_SESSION : 0bd3ab18: Media Session 412 has been created
SIP 14:58:59.727: USER_AGENT : Added Media session(412) [callId="AICCCB9F4C51812D18_00:08:5d:9a:52:76", localTag="AIE24EDCC980177CC0", remoteTag=""] to SessionIdMap sessP = 0bd3ab18
SIP 14:58:59.727: USER_AGENT : Added Media session(412) [callId="AICCCB9F4C51812D18_00:08:5d:9a:52:76", localTag="AIE24EDCC980177CC0", remoteTag=""] to DialogIdMap sessP = 0bd3ab18
SIP 14:58:59.728: UA_MEDIASESS: 0bd3ab18: TrunkSessionC::handleEstablishOutgoingCall(Bcs: 412), transitCounter: 255
SIP 14:58:59.729: UA_MEDIASESS: 0bd3ab18: Bcs: 412 Preferred Pty Name is already converted to UTF8 'Gudeco Nuernberg'
SIP 14:58:59.729: UA_MEDIASESS: Session (Bcs: 412) sets P-Asserted-Identity to displayName: [Gudeco Nuernberg] user: [+4991153992358], host: [sip-trunk.telekom.de]
SIP 14:58:59.729: UA_MEDIASESS: 0bd3ab18: handleEstablishOutgoingCall(Bcs: 412), from: ["Gudeco Nuernberg" <sip:[email protected]>], to: [sip:[email protected]]
SIP 14:58:59.729: UA_MEDIASESS: 0bd3ab18: Media Session Bcs: 412 changed state from 'Idle' to 'Connecting'
SIP 14:58:59.729: UA_MEDIASESS: setConnectedPartyName("Gudeco Nuernberg" ), myConnectedPtyName: []
SIP 14:58:59.729: UA_MEDIASESS: 0bd3ab18: Bcs: 412 Connected Pty Name is already converted to UTF8 'Gudeco Nuernberg'
SIP 14:58:59.730: UA_MEDIASESS: lineUpMediaSections(Bcs: 412)
SIP 14:58:59.730: UA_MEDIASESS: - mediaInfo: [Offer: 10.10.0.11;voip[1]=5022(sendrecv)]
SIP 14:58:59.730: UA_MEDIASESS: - myRemoteMediaInfo: [Empty]
SIP 14:58:59.730: UA_MEDIASESS: - MediaSectionList(Bcs: 412): Audio: [-1], Video: [-1], TCPImage: [-1], UDPTLImage: [-1], Unknown: [-1]
SIP 14:58:59.730: UA_MEDIASESS: - Updated MediaSectionList: Audio: [1], Video: [-1], TCPImage: [-1], UDPTLImage: [-1], Unknown: [-1]
SIP 14:58:59.730: UA_MEDIASESS: setSDPFromMediaInfoBcs: 412), constructOffer: true
SIP 14:58:59.730: UA_MEDIASESS: - mediaInfo: [Offer: 10.10.0.11;voip[1]=5022(sendrecv)]
SIP 14:58:59.730: UA_MEDIASESS: setAudioInSDP(): media: Offer: 10.10.0.11;voip[1]=5022(sendrecv)
SIP 14:58:59.731: UA_MEDIASESS: updateCodecListIfRequired : Endpoint 65537 has codec: Undef and Codec Mode=Off
SIP 14:58:59.731: UA_MEDIASESS: updateCodecListIfRequired : Endpoint 65537, Codec mode is unspecified, no changes done to codecList.
SIP 14:58:59.732: UA_SESSION : sendSipMessage(): myPriRemoteAddr: 217.0.26.101:5060:TCP
SIP 14:58:59.732: UA_SESSION : 0bd3ab18: sendSipMessage(): Sending a request to clientId [] / sipEpId [1(SipProvider)]
SIP 14:58:59.732: UA_SESSION : 0bd3ab18: Session has myAuthMode: AuthOff
SIP 14:58:59.732: UA_SESSION : 0bd3ab18: getNextHop(), myPriRemoteAddr: 217.0.26.101:5060:TCP
SIP 14:58:59.732: UA_SESSION : 0bd3ab18: getNextHop(), returning nextHopAddr: 217.0.26.101:5060:TCP
SIP 14:58:59.733: UA_SESSION : 0bd3ab18: Sending SIP message to 217.0.26.101:5060:TCP, nextHopAddr=[217.0.26.101:5060:TCP], myPriRemoteAddr=[217.0.26.101:5060:TCP], mySecRemoteAddr=[0.0.0.0:0:UDPorTCP]
SIP 14:58:59.733: USER_AGENT : IaSipUserAgentC::sendSipMessage(): Sending a message to clientId [] / sipEpId [1(SipProvider)]
SIP 14:58:59.734: USER_AGENT : sending message to remoteNodeP: 0 / providerId: 1
SIP 14:58:59.734: USER_AGENT : Setting Route headers in message, realmName = [sip-trunk.telekom.de]
SIP 14:58:59.735: USER_AGENT : IaSipUserAgentC::sendSipMessage. providerId != NULL
SIP 14:58:59.735: USER_AGENT : IaSipUserAgentC::adaptContactUriForAccount.Failed to find an account!!!
SIP 14:58:59.736: UA_SIP_MSG : LAN < UA: "INVITE" request [1 INVITE] sent to [217.0.26.101:5060:TCP]
SIP 14:58:59.736: UA_SIP_MSG : INVITE sip:[email protected] SIP/2.0
SIP 14:58:59.736: UA_SIP_MSG : Via: SIP/2.0/TCP 10.10.0.10;branch=z9hG4bK_AI2022May1158597330821207124578242;rport
SIP 14:58:59.736: UA_SIP_MSG : To: sip:[email protected]
SIP 14:58:59.736: UA_SIP_MSG : From: "Gudeco Nuernberg" <sip:[email protected]>;tag=AIE24EDCC980177CC0
SIP 14:58:59.736: UA_SIP_MSG : Call-ID: AICCCB9F4C51812D18_00:08:5d:9a:52:76
SIP 14:58:59.736: UA_SIP_MSG : CSeq: 1 INVITE
SIP 14:58:59.736: UA_SIP_MSG : Route: <sip:reg.sip-trunk.telekom.de:5060;lr>
SIP 14:58:59.736: UA_SIP_MSG : Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,PUBLISH,UPDATE,REFER,PRACK
SIP 14:58:59.736: UA_SIP_MSG : Allow-Events: presence,dialog,message-summary,refer
SIP 14:58:59.736: UA_SIP_MSG : Max-Forwards: 70
SIP 14:58:59.736: UA_SIP_MSG : User-Agent: Aastra 400
SIP 14:58:59.736: UA_SIP_MSG : Content-Type: application/sdp
SIP 14:58:59.736: UA_SIP_MSG : Privacy: none
SIP 14:58:59.736: UA_SIP_MSG : Accept: application/sdp
SIP 14:58:59.736: UA_SIP_MSG : P-Asserted-Identity: "Gudeco Nuernberg" <sip:[email protected]>
SIP 14:58:59.736: UA_SIP_MSG : P-Early-Media: supported
SIP 14:58:59.737: UA_SIP_MSG : Supported: 199,100rel
SIP 14:58:59.737: UA_SIP_MSG : Contact: <sip:[email protected]:5060;transport=tcp>
SIP 14:58:59.737: UA_SIP_MSG : Content-Length: 244
SIP 14:58:59.737: UA_SIP_MSG :
SIP 14:58:59.737: UA_SIP_MSG : v=0
SIP 14:58:59.737: UA_SIP_MSG : o=aastra400 1099359712 1099359712 IN IP4 10.10.0.10
SIP 14:58:59.737: UA_SIP_MSG : s=call
SIP 14:58:59.737: UA_SIP_MSG : c=IN IP4 10.10.0.11
SIP 14:58:59.737: UA_SIP_MSG : t=0 0
SIP 14:58:59.737: UA_SIP_MSG : m=audio 5022 RTP/AVP 8 0 101
SIP 14:58:59.737: UA_SIP_MSG : a=rtpmap:8 PCMA/8000
SIP 14:58:59.737: UA_SIP_MSG : a=rtpmap:0 PCMU/8000
SIP 14:58:59.737: UA_SIP_MSG : a=rtpmap:101 telephone-event/8000
SIP 14:58:59.737: UA_SIP_MSG : a=fmtp:101 0-15
SIP 14:58:59.737: UA_SIP_MSG : a=sendrecv
SIP 14:58:59.737: UA_SIP_MSG : a=ptime:20
SIP 14:58:59.737: USER_AGENT : IaSipUserAgentC::sendSipMessage. MTU flag [FALSE]
SIP 14:58:59.738: TRANSCEIVER : transmitSipMessage(217.0.26.101:5060:TCP)
SIP 14:58:59.738: TRANSCEIVER : updateHostPort(Command,217.0.26.101:5060:TCP)
SIP 14:58:59.739: TRANSCEIVER : updateHostPort() did not update the destination address [217.0.26.101:5060]
SIP 14:58:59.739: TRANSCEIVER : transmitSipMessage() has set the sendAddress to 217.0.26.101:5060
SIP 14:58:59.739: TRANSCEIVER : Outgoing SIP message was not sent through the ALG
SIP 14:58:59.739: TRANSCEIVER : transmitSipMessage() changed the transport to 'TCP'
SIP 14:58:59.739: TRANSCEIVER : LAN < UA: "INVITE" request [1 INVITE] sent to [217.0.26.101:5060:TCP]
SIP 14:58:59.739: TRANSCEIVER : INVITE sip:[email protected] SIP/2.0
SIP 14:58:59.739: TRANSCEIVER : Via: SIP/2.0/TCP 10.10.0.10;branch=z9hG4bK_AI2022May1158597330821207124578242;rport
SIP 14:58:59.739: TRANSCEIVER : To: sip:[email protected]
SIP 14:58:59.739: TRANSCEIVER : From: "Gudeco Nuernberg" <sip:[email protected]>;tag=AIE24EDCC980177CC0
SIP 14:58:59.739: TRANSCEIVER : Call-ID: AICCCB9F4C51812D18_00:08:5d:9a:52:76
SIP 14:58:59.739: TRANSCEIVER : CSeq: 1 INVITE
SIP 14:58:59.739: TRANSCEIVER : Route: <sip:reg.sip-trunk.telekom.de:5060;lr>
SIP 14:58:59.739: TRANSCEIVER : Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,PUBLISH,UPDATE,REFER,PRACK
SIP 14:58:59.739: TRANSCEIVER : Allow-Events: presence,dialog,message-summary,refer
SIP 14:58:59.739: TRANSCEIVER : Max-Forwards: 70
SIP 14:58:59.739: TRANSCEIVER : User-Agent: Aastra 400
SIP 14:58:59.739: TRANSCEIVER : Content-Type: application/sdp
SIP 14:58:59.739: TRANSCEIVER : Privacy: none
SIP 14:58:59.739: TRANSCEIVER : Accept: application/sdp
SIP 14:58:59.739: TRANSCEIVER : P-Asserted-Identity: "Gudeco Nuernberg" <sip:[email protected]>
SIP 14:58:59.740: TRANSCEIVER : P-Early-Media: supported
SIP 14:58:59.740: TRANSCEIVER : Supported: 199,100rel
SIP 14:58:59.740: TRANSCEIVER : Contact: <sip:[email protected]:5060;transport=tcp>
SIP 14:58:59.740: TRANSCEIVER : Content-Length: 244
SIP 14:58:59.740: TRANSCEIVER :
SIP 14:58:59.740: TRANSCEIVER : v=0
SIP 14:58:59.740: TRANSCEIVER : o=aastra400 1099359712 1099359712 IN IP4 10.10.0.10
SIP 14:58:59.740: TRANSCEIVER : s=call
SIP 14:58:59.740: TRANSCEIVER : c=IN IP4 10.10.0.11
SIP 14:58:59.740: TRANSCEIVER : t=0 0
SIP 14:58:59.740: TRANSCEIVER : m=audio 5022 RTP/AVP 8 0 101
SIP 14:58:59.740: TRANSCEIVER : a=rtpmap:8 PCMA/8000
SIP 14:58:59.740: TRANSCEIVER : a=rtpmap:0 PCMU/8000
SIP 14:58:59.740: TRANSCEIVER : a=rtpmap:101 telephone-event/8000
SIP 14:58:59.740: TRANSCEIVER : a=fmtp:101 0-15
SIP 14:58:59.740: TRANSCEIVER : a=sendrecv
SIP 14:58:59.740: TRANSCEIVER : a=ptime:20
SIP 14:58:59.740: TRANSCEIVER : IaSipTransceiverC::transmitSipMessage() encoding the message
SIP 14:58:59.740: TRANSPORT : IaSipTransportC::send:217.0.26.101:5060:TCP
SIP 14:58:59.740: TRANSPORT : IaSipTransportC::sendOverTcp
SIP 14:58:59.740: TRANSPORT : Found open TCP socket: [0xbfa90c8]
SIP 14:58:59.740: TRANSPORT : [0xbfa90c8]IaSipTcpSocketC::sendData(217.0.26.101:5060:TCP)
SIP 14:58:59.741: TRANSPORT : IaSipTcpSocketC::sendPendingData
SIP 14:58:59.741: TRANSPORT : [0xbfa90c8]IaSipSocketC::sendData:217.0.26.101:5060:TCP
SIP 14:58:59.741: TRANSPORT : Data sent (ok to send more)
SIP 14:58:59.741: TRANSPORT : IaSipTcpSocketC::startInactivityTimer
SIP 14:58:59.741: TRANSPORT : Started inactivity timer
SIP 14:58:59.747: TRANSPORT : [0xbfa90c8]IaSipTcpSocketC::handleSocketData:217.0.26.101:5060:TCP
SIP 14:58:59.747: TRANSPORT : IaSipTcpSocketC::startInactivityTimer
SIP 14:58:59.747: TRANSPORT : Started inactivity timer
SIP 14:58:59.747: TRANSPORT : IaSipTransportC::handleData:217.0.26.101:5060:TCP
SIP 14:58:59.747: TRANSCEIVER : LAN > UA: "100 Trying" status [1 INVITE] received from [217.0.26.101:5060:TCP]
SIP 14:58:59.747: TRANSCEIVER : SIP/2.0 100 Trying
SIP 14:58:59.748: TRANSCEIVER : Via: SIP/2.0/TCP 10.10.0.10;rport;received=212.185.87.218;branch=z9hG4bK_AI2022May1158597330821207124578242
SIP 14:58:59.748: TRANSCEIVER : To: <sip:[email protected]>
SIP 14:58:59.748: TRANSCEIVER : From: "Gudeco Nuernberg" <sip:[email protected]>;tag=AIE24EDCC980177CC0
SIP 14:58:59.748: TRANSCEIVER : Call-ID: AICCCB9F4C51812D18_00:08:5d:9a:52:76
SIP 14:58:59.748: TRANSCEIVER : CSeq: 1 INVITE
SIP 14:58:59.748: TRANSCEIVER : Content-Length: 0
SIP 14:58:59.748: TRANSCEIVER :
SIP 14:58:59.748: TRANSCEIVER : Incoming SIP message was not sent through the ALG
SIP 14:58:59.748: UA_SIP_MSG : LAN > UA: "100 Trying" status [1 INVITE] received from [217.0.26.101:5060:TCP]
SIP 14:58:59.749: UA_SIP_MSG : SIP/2.0 100 Trying
SIP 14:58:59.749: UA_SIP_MSG : Via: SIP/2.0/TCP 10.10.0.10;rport;received=212.185.87.218;branch=z9hG4bK_AI2022May1158597330821207124578242
SIP 14:58:59.749: UA_SIP_MSG : To: <sip:[email protected]>
SIP 14:58:59.749: UA_SIP_MSG : From: "Gudeco Nuernberg" <sip:[email protected]>;tag=AIE24EDCC980177CC0
SIP 14:58:59.749: UA_SIP_MSG : Call-ID: AICCCB9F4C51812D18_00:08:5d:9a:52:76
SIP 14:58:59.749: UA_SIP_MSG : CSeq: 1 INVITE
SIP 14:58:59.749: UA_SIP_MSG : Content-Length: 0
SIP 14:58:59.749: UA_SIP_MSG :
SIP 14:58:59.749: USER_AGENT : Media Session 0bd3ab18 found for incoming STATUS where CSeq="1 INVITE"
SIP 14:58:59.749: UA_SESSION : IaSipUaBaseSessionC::checkSupported.
SIP 14:58:59.749: UA_SESSION : IaSipUaMediaSessionC::checkSupported() SR ENabled in monitor
SIP 14:58:59.750: UA_MEDIASESS: 0bd3ab18: handleSipStatus100(Bcs: 412) in state: Connecting, myOrigSessionState: Invalid
SIP 14:58:59.750: UA_SESSION : 0bd3ab18: BaseSession::handleSipStatus() in state: Connecting
SIP 14:58:59.750: UA_SESSION : 0bd3ab18: Status 100 received for "INVITE" request
SIP 14:58:59.750: UA_SESSION : 0bd3ab18: thisSeqNum=1, myLastHandledStatusSeqNum=0, myLastHandledStatusCode=0
SIP 14:58:59.750: UA_MEDIASESS: 0bd3ab18: Status 100 received for "1 INVITE" request for Bcs: 412
SIP 14:58:59.750: UA_MEDIASESS: Bcs: 412, Checking remote pty nr/name in SipStatus: 100 for INVITE in state: Connecting
SIP 14:58:59.750: UA_MEDIASESS: setRemotePtyInfo(), currently: myRemoteNr: [], myRemoteName: []
SIP 14:58:59.750: UA_MEDIASESS: Changed, myRemoteNr: [0821207124578], myRemoteName: []
SIP 14:58:59.756: ER-PBX_IFACE: ER > PBX: CallProc(Bcs: 412), pref=27
SIP 14:58:59.756: ER_S_SESSION: Call Bcs: 412: Changed state from Idle to Proceeding
SIP 14:58:59.756: ER_S_CALLHDL: ER < PBX: indCallProceeding(Bcs: 411), progInd: 0, cdNr: 0-0821207124578, cdName:
IBT 14:58:59.756: BCS: Client: tpCtlProc Bcs: 411 MT: Reserve
SIP 14:58:59.758: ER-PBX_IFACE: ER < PBX: Call Proceeding(Bcs: 411) received progInd: 0, cdNr: 0-0821207124578, cdName:
SIP 14:58:59.758: ER_S_SESSION: Session(Bcs: 411) in state: Initiate, CallProceeding from PBX
SIP 14:58:59.758: ER_S_SESSION: - cdPtyNb: [0-0821207124578], cdPtyName: []
SIP 14:58:59.759: ER_S_SESSION: Call Bcs: 411: Changed state from Initiate to Proceeding
SIP 14:58:59.760: USER_AGENT : IaSipUserAgentC::sendSipMessage(): Sending a message to clientId [TermId:15] / sipEpId [None]
SIP 14:58:59.760: USER_AGENT : sending message to client: TermId:15
SIP 14:58:59.760: USER_AGENT : NO Route header added for provId=0, msgType= not STATUS
SIP 14:58:59.760: USER_AGENT : IaSipUserAgentC::sendSipMessage. providerId == NULL
SIP 14:58:59.760: USER_AGENT : IaSipUserAgentC::adaptContactUriForAccount provider not found.
Zum Vergrößern anklicken....
 
Zuletzt bearbeitet von einem Moderator:
Dein Log ändert auf den ersten Blick mit einem 100 Trying. Was danach kommt, wäre interessant.

Die Anzahl deiner Kanäle findest du a) in deinen Telekom Unterlagen und b) in der Anlage unter Systemübersicht - Lizenzierung und dort unter Network - SIP Access Channels. Beide Angaben sind wichtig.
 
@JD168
Bei mehrzeiligen Logs bitte unbedingt [CODE] Tags verwenden!
 
Hab es mal in QUOTE gesetzt, da jetzt natürlich aus den eMailadressen im BBCode EMAIL wurde... die Zeit und Lust habe ich nicht die Stellen rauszulöschen
 
In der Anlage unter Systemübersicht - Lizenzierung unter Network - SIP Access Channels steht 54. Bedeutet dass, das 54 Kanäle zur Verfügung stehen oder aktuell in Benutzung sind?

Die Logdatei von gestern liegt mir aktuell nicht mehr vor. Muss auf eine neue Meldung warten. Wie schauen denn Fehlermeldungen in diesem Log generell aus? Anruf misslungen = Service unavailable = Kanäle zu wenig?
 
Das sind die Kanäle die dir in der Anlage zur Verfügung stehen. Aktuell belegte Kanäle kannst du in der Systemübersicht - Status - System - Besetzte externe B-Kanäle sehen.

chrsto schrieb:
Die Anzahl deiner Kanäle findest du a) in deinen Telekom Unterlagen
Zum Vergrößern anklicken....

Das hast du noch nicht beantwortet. Wie viele Kanäle hast du bei der Telekom gebucht?

Wie die Fehlermeldung aussieht, weiß ich nicht, da wir noch nicht wissen, was das für ein Fehler ist. Dafür wäre ein Mitschnitt im Fehlerfall hilfreich.
 
Telekomvertrag sind 30 gelistet:
Besetzte interne B-Kanäle5
Besetzte externe B-Kanäle1

Fehlermeldung muss ich auf die Rückmeldung des Anwenders warten...
 
Um antworten zu können musst du eingeloggt sein.

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